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Pa Guide

Pa Guide

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Issue 1.0 May 1999 Written by Mike Holman

Although we are aware of the fact that the vast majority of our users would rather not have to learn anything about Professional Audio, we are also aware that it is a necessary evil. We offer this document for reference by those seeking more information. It is by no means a finished work. Nor does it cover all topics necessary. However, it is a starting point. We hope that you can make use of it and encourage your feedback. If it gets too technical in spots, we encourage you to keep moving and read the document several times. We also encourage you to print this series of documents out and keep them for future reference. For those of you who are looking for something real basic, check out our “P.A. Basics”.

System Setup Running the System Signal Flow Microphones Sources Mixer
Mixer Inputs Mixer Outputs Mixer Controls

Page Number
3 6 8 9 12 13 13 15 16 22 27 34 34 37 39 42 44

Signal Processors Power Amplifiers Speakers Basics and Terminology Cable Issues Power and SPL Phase and Frequency Components and Enclosure

Professional Audio Basics and User Tips Guide


GETTING STARTED The following is a simplified set of instructions for general applications. The aim is to get a basic system up and running with as few problems as possible. [1] A BASIC STAGE SETUP GOES AS FOLLOWS: • (a) Place the front monitors near the edge of the stage, aiming back at the performers. • (b) Place the mics and stands in front of their respective monitors at a distance of 1 to 3 feet, depending on the cabinet's up-facing angle - the speakers should be aiming directly at the backs of the mics. Other monitors should be located as closely as possible to the performers, also aiming at the backs of their mics. This is to reduce feedback potential. • (c) Place the main FOH (front-of-house) speakers at stage front at the far corners aiming straight out at the audience. Do not aim them in at the audience in front of centre stage unless the stage is deep enough for the mics to be set farther back to reduce feedback. { TIP - If stage-front-centre audience coverage is a problem, perhaps because they are too close to the stage to hear the FOH speakers, try turning a spare monitor around to face them}. [2] CONNECTIONS GO AS FOLLOWS: • (a) Connect all mics, line-level signal sources - i.e., tape decks, CD players, instrument amp line outputs, etc.- also processors and external effects units, to their respective channel inputs or send and return jacks on the mixer (for more information see INPUTS and OUTPUTS under THE MIXER). { TIP - It is also a good idea to identify the various channel sources, perhaps with small stick-on labels at the bottoms of the channels, e.g., "lead vocal", "guitar", "drum vocal", etc.} • • (b) Do not connect speakers yet. Transients from things being plugged into the mixer and switched on can cause speaker damage eventually if not immediately. Connect speakers last. (c) Similarly, when you are connecting one or more external power amplifiers and speakers, be sure to connect the speakers to the power amp after it has been connected to the mixer and the mixer has been powered up. The reason for this is because some mixers do not have turn-on transient suppression built in. Any power amp(s) which are connected to the mixer and running with their speakers connected when the mixer is switched on can amplify this large burst of signal voltage with a resounding "pop" or "boom" and the speaker system may be damaged. Even if the speakers survive this type of accident the first time, repeated accidents will eventually take their toll. { TIP - If there is a power failure during the job, try to switch off the power amps immediately, before the power comes on again. Mixers with built-in power amps usually don't suffer from "pop" problems when switched on, but it's not a bad idea to turn the master levels off before powering them up.} • (d) Electronic crossovers or speaker processors should be connected between the mixer and power amplifiers. The term "between" means mixer output to unit's input and unit's output to power amp's input(s). There are four types of processors which you may encounter. First, there are simple processors which provide pre-equalization for specific speaker systems. They are connected between the mixer and power amp(s) in the same manner as an equalizer. Secondly there are adjustable active crossovers which are connected in a similar manner except their outputs must go to separate power amps or amp channels each driving the appropriate woofers, horns and tweeters, or subwoofers and full-range enclosures.

Professional Audio Basics and User Tips Guide


Remember to find out the speaker manufacturer's recommended crossover frequencies for the various elements in order to set the crossover accurately. If this is not possible, be careful when doing this by "ear". What sounds right to you may be wrong for the components' long-term reliability. To be safe, set the output level controls all at maximum so that they are at the same output level (at lower settings, potentiometer tolerances could cause them to be different) then counter-adjust (lower) the crossover's input level to avoid over-driving the power amps. The third type of processor is a combination crossover/processor. This would be connected and employed as if it were a crossover, although such units tend to be for specific speaker systems and therefore do not usually have variable crossover frequencies. Now a fourth type of unit has recently emerged from the depths of techno-wizardry to test our connecting skills. The "sensing" processor / crossover / compressor is designed for use with specific speakers in a touring system. In this case, everything gets hooked up as if it were a just a crossover - with one exception, additional speaker cables are run from the speaker outputs on the power amps to "sensing" inputs on the processors. Then, when the system is up and running with the amps set at full output, you push the processor's "calibrate" buttons. A set of tones is sent through the amplifiers and speakers while the "sense" circuitry samples some of the amplifier's output signal and programs the processor's variable parameters such as crossover frequency and gain. [3] THE SOUND CHECK It is very important to do a sound check before the job starts. It is also important that the band is playing during the sound check. However if this is not possible, have someone test the mics while you adjust the mixer. • (a) With the master levels turned off, adjust the channel input gain or attenuator settings so that the input "clip" indicators flash. Now turn them down slightly. This ensures that adequate amounts of source signal are driving the channel circuitry to provide an optimum signal-to-noise ratio. • (b) If the mixer does not have input gain/atten. controls or input clip indicators, it is a good idea to turn up the channel level or volume controls around one-third before turning up the master levels. • (c) Now bring up the masters slowly and re-adjust the various channel level controls for the desired mix through the FOH PA. • (d) When the sound check is finished, remember to turn off channels which only get used once in a while - e.g. harmonica mic channel, acoustic guitar channel, special percussion instrument mic channel, etc. This is to prevent open mics from causing feedback and noise problems. Channel mute buttons are handy for this function (see >>MUTE under THE MIXER). Otherwise you can use a washable marker to mark the settings before turning these levels down. • (e) Generally speaking, channel EQ controls should be kept as close to flat (centre position) as possible, especially on mic channels. In any case, start the soundcheck with all channel EQ controls at centre. Adjustments should be made for specific purposes - remember, this is a PA system, not a home stereo (amoung other things, home stereos don't have to produce 110 to 130dB SPLs with zero feedback). Possible channel EQ settings could be as follows; [ vocals - all flat ] [ acoustic guitar - bass flat, mid -3dB, treble +2dB ] [ bass drum - bass flat, mid -3dB, treble flat ] [ snare drum - bass -6db, mid & treble - flat ] [ floor toms - all flat ] [ tenor toms - bass -3dB, mid & treble - flat ] [ cymbals - bass -12dB, mid & treble - flat ] [ trumpet, tenor & alto saxophone - bass -3dB, mid & treble flat ] [ trombone, baritone saxophone- all flat ] [ guitar amp - bass & mid flat, treble -3dB ] [ bass amp - bass flat, mid -3db, treble flat ] [ harmonica - bass -6dB, mid flat, treble -3dB ]

Professional Audio Basics and User Tips Guide


but aside from that. (g) Final monitor or aux. remember that reverb or echo should be added to the monitors in smaller amounts than to the FOH system to avoid feedback. You could do the same with the monitor master(s) and EQ. for instance. The downside to this practice is that the system's response is now EQ'd sufficiently to sound strange. To get started however. assume that the vocal channels will need to be loudest through the monitors. and the rest of the band to the right submaster thus creating two submixes to facilitate level adjustments (hopefully the mixer will have a MAIN. or multiple monitor systems could be adjusted similarly at first then re-adjusted according to the various artists' wishes. (j) If you're playing a new hall. Then you increase the levels some more and re-adjust the main EQ. Drum mic channel monitor sends may be left off at first. (h) If there are effects-to-monitor masters.e. keyboards and horns can sometimes also use small amounts of either effect. send levels should be set according to the artists' needs. The only time you might set them differently would be if you decided to run a mono PA.• • • • • (f) Reverb or echo should be added principally to vocals and in small amounts. and it may get worse because the house acoustics will change when the audience is in place. i. The process is simply a matter of increasing the main levels until you get feedback then EQing it away (see >>EQ under THE MIXER (Master Section Controls)). but the band is NOT likely to appreciate it. SUM OR MONO under MIXER OUTPUTS). the channel pan controls should be set at center position. but drums and bass guitar should usually be kept "dry" to allow a firm-sounding foundation for the rhythm section. to the left submaster. Professional Audio Basics and User Tips Guide 5 . Lead guitar. other active channels should have lower monitor settings. SUM or MONO buss and master to permit this). then turned up as required. Then you could use the pans to send all the drum mic channels. perhaps for keyboards or drums. (i) If you are using a stereo mixer. acoustic guitar will also require a fairly high monitor send setting. with the main power amplifiers all inter-connected to a summed mono output on the mixer (see >>MAIN. A better practice is to position the FOH speakers so that feedback even at high volume levels is minimized. you might consider pre-equalizing the system against possible future feedback problems. A secondary monitor system.

although low-frequency hearing can be affected as well. the highs will be the first to run out of headroom and distort. for example. backup vocals and seldom-used channels that are kept shut off until needed. and roughly the same ones as above. there is an increased risk of high-frequency feedback and. feedback should not become a problem unless someone gets too close to a speaker with a mic. a harmonica mic or acoustic guitar mic (did you do an "oops" and leave it on?). should you need to boost the FOH level significantly later on. You're better off to leave the EQ or crossover alone. you should not have to ride the channel levels very much. you will have to bring that mic channel's level and monitor level up . a trick which should cause them to compensate by singing louder and/or getting closer to the mic (just remember to reward them by bringing their monitor level back up). is probably extra loud as well. If that works. by then. Some "usual suspects" include: • • • Overhead drum/cymbal mics may be suspect if the drummer has a monitor. As a result. when the harmonica player is ready to play.RUNNING THE SYSTEM LEVEL MAINTENANCE If you have done all the preparatory setup work during a sound check. you'll have to make your best guess. A singer's acoustic guitar mic might be too close to a monitor.see SETUP section 3 item (a) . Earplugs are a worthwhile investment. MONITOR LEVELS Channel monitor level settings will require some adjustments as well. Most common amoung these is loss of high-frequency sensitivity. FATIGUE During the job.the light on the channel which is feeding back should be brighter and on more steadily than it was before. which would normally be left off until needed. When that happens. Your hearing should normalize by the next day. try the following. Lead vocals may need slight level adjustments if the vocalist has a habit of fading back. e. If they persist in doing this. you can probably see which mic it is and know which channel's monitor or PA level to turn down. (A) If you have set the input channel gains high enough for there to be some clip light activity . The only levels you should need to change would be instrument solos. pre-EQ monitor sends represent an independent mix. turn down the channel's monitor send level and if it's too close to the FOH speakers pull down the channel's level fader. they hear an unpleasantly bright sound which. In this case turn down the monitor sends and pull down the channel fader. This is NOT a good idea. etc.that the horns or tweeters have somehow run out of "steam". So. Scan the Clip LED's and turn down that channel's monitor send level. possibly damaging horns and/or tweeters. a possible solution may be to turn that individual down through the monitors. people often assume that the speaker system has lost some high end .ditto for horn mic channels. Too often there are customers who have just come in from somewhere else and their hearing is not fatigued.g. If it's too close to a monitor. But. Try turning down the mic channels' monitor sends. any clip light action at all would be an indicator. Since there are no other symptoms such as discomfort or ringing in the ears. acoustic guitar mic channels. you will probably suffer from symptoms of hearing fatigue to some degree. if it's not immediately obvious which mic is responsible. Additionally. If it's a currently unused mic whose channel should be shut off. DEALING WITH FEEDBACK Assuming everything is done right during the setup and sound check. Keep in mind that pre-fader. but don't deaden the sound too much. if you didn't set the gains high enough to use the clip lights. Professional Audio Basics and User Tips Guide 6 . However. Their next reaction is often to boost the high-frequency EQ or the HF gain on the crossover. try using the channel EQ to minimize feedback.

Now ease up the master. • { TIP . to turn off the amp's volume control.} (B) The average audience member or performer will put up with feedback about that (snap your fingers) long. And further suspects could be: • a stuck keyboard note.or whatever. Of course if lowering the monitor master levels does not work you'll need to lower the main masters. go to your main or monitor EQ.with flattops.If a guitarist or singer/guitarist insists on leaving their standby electric-acoustic guitar plugged into an amp with it turned on and the volume left up. you can accomplish the same feat by taking one (only) of the offending mic's cable connectors apart and reversing the leads. Someone onstage will have to turn down the offending volume control or free-up the stuck key or move the mic . (E) There may actually be situations where pulling down the main AND monitor masters fails to end the howling completely. • a clavinet or other stringed keyboard instrument and keyboard amp feeding back. A better solution. Even if your mixer does not have phase buttons. mistakenly left on and feeding back into it. try (at least) to encourage them to leave their pick in the strings that resonate. If the feedback starts again. Likely suspects would include feedback from a spare electric guitar and amp or an electric/acoustic guitar and amp waiting to be used and mistakenly left on. Eventually. you'll have it under control. but it needs to be done.you need to normalize the EQ as much as possible. This can be done later. is for them to leave the guitar's volume control turned off or. If your mixer has input phase reversal buttons. Try turning down the guitar mic channel's monitor send. Putting a channel or source signal out of phase does not affect the sound. Any of these problems are impossible to cure from the mixing station. but now the main or monitor system frequency response has been altered and probably doesn't sound right. Pull down a few of the sliders slightly (-3dB) in the frequency range which your ears tell you is likely to be the right one. but it requires knowing which string(s) to deaden. Bring the monitor masters back up to a point below where they were before so they can have some coverage. It works when the problem is that two mics are picking up the same source and feeding it into a monitor close to those mics. the offending source signal riding on that channel gets cancelled out by the similar signal in the other channel and the problem is solved. Hint . only much more timeconsuming and it might even cause new feedback problems later on. Professional Audio Basics and User Tips Guide 7 . (C) Now it is important to find a quick remedy which will let you get the main or monitor levels back up to where they should be.Since vocal mics tend to pick up some flattop and that gets into the monitors along with the actual guitar mic signal which singers usually want to hear at a goodly level. DO NOT PULL DOWN ALL THE EQ FADERS AT ONCE. lower the master a little. if it doesn't have one. First. and hopefully soon. A or D string. it's an invitation to feedback. re-centre the EQ faders you just pulled down and try pulling down some other frequencies then bringing the master back up. By putting one (only) of those two channels out of phase. You still need to find these two culprits then re-position them or insert an EQ in that mic's channel in order to solve the problem properly. If the problem isn't solved by now. of course. go to plan B .Mixer input phase reversal is sometimes a very effective way to get rid of certain persistent feedback problems. It's a simple solution.haul down the monitor master(s). try reversing the phase on the guitar mic channel so you can bring the monitor level back up without feedback. (D) The problem remains that a mic and a speaker have decided to feed back. • { TIP .}. That should solve the problem. • an unused instrument mic plugged directly into a powered monitor or combo amp. it's most often the low E. Try carefully pushing some of the EQ sliders back up towards centre position . That would be about the same thing as lowering the mixer masters. but now the band has no monitors.

But the microphone represents both the most commonly used signal source and the one which requires the most understanding. EQ. power amplifiers and preamp devices this is called headroom and in speakers it's called power capacity. a CD player. a tape deck. etc. At all times the basic signal path is from A to B to C. the Line Out from an amp.reverb. think of electrons. help you to prevent distortion by telling you when a circuit in a mixer. the details will probably make more sense.SIGNAL FLOW It is beneficial to start looking at PA from the standpoint of how signals move within the system. "Clip" indicators. The source can be any one of many different things from an electric instrument to the line-level output of an instrument amplifier. This A>B>C concept is worth keeping in mind because each stage after the signal source has a certain maximum capacity. an electric instrument.or a power amplifier is on the verge of being overloaded. such as LED's. They simply distort and/or self-destruct unless they have fuses or circuit breakers built in. Try forcing too many electrons into something and you exceed that capacity. Professional Audio Basics and User Tips Guide 8 . the output of another mixer or basically anything which produces an audio signal • [B] The amplification stage which includes the preamplification (mixer) stage • [C) The speaker system. The signal source produces a small quantity of electrons. That way. The amplification stage then adds many more and the speaker utilizes this enlarged amount of electrons to create the physical motion which produces sound. To make the term "signal" a little more graphic. In mixers. signal processor . the PA process begins with a source signal. . As mentioned above. a tape deck or CD player or the output of another mixer. A>B>C A PA system contains three basic elements: • [A] The signal source which can be a microphone. usually with audible results such as distortion or blown speakers. Unfortunately there is nothing similar for speakers.

Certain models are also preferred for bass drum and others for brass instruments. piano or anything with strings. mandolin. or from the mixer if it has phantom power built in. It works rather like a speaker in that there is a diaphragm attached to a coil of hair-thin insulated wire flexibly suspended in a magnetic field. Condenser mic technology is ideal for virtually all applications with the possible exception of bass drum. "Directionality" or "Polar Response") Most microphones are capable of picking up sounds approaching from a wide area. A third type. to the line input on a mixer and hook the mic up to the amplifier outputs. the bigger the waves and the farther the coil moves hence cutting more lines of magnetic force and generating more voltage). Dynamic mics are best for close-up use whether for vocals. This is deposited as positive and negative charges on two thin metal plates with a small airspace or other resistive material between them. thus a small amount of voltage is induced in the coil. Although the polar plot diagram is flat-looking. • Pickup patterns can be imagined as invisible balloons. It acts as a dielectric and a signal voltage is produced that varies in polarity and amplitude with the frequency and amplitude of the sound waves. Talk into the speaker and sound will come out of the mic. In fact. They operate on a small amount of DC voltage either from a built-in battery or a "phantom" power supply unit. The all-important midrange and high frequency sounds approaching from outside a mic's pickup pattern will be detected at far lower sound pressure levels those which are approaching from within the pattern and will get drowned out by them. This voltage travels down the mic cable to the mixer where it is amplified and sent to the speaker.MICROPHONES BASIC TYPES There are two basics types of microphone commonly used in pro audio. banjo. the "ribbon" mic is sufficiently fragile that they seldom get used in live music situations. PICKUP PATTERNS (a. CONDENSER MICROPHONES Condenser microphones offer high sensitivity and smooth frequency response. a woofer for example. choirs for example. in reality mics pick up sounds coming from above and below as well as the front and sides and even the back. They are also preferred for overhead coverage of drum sets. DYNAMIC MICROPHONES The dynamic mic is most commonly found in PA applications due to its general ruggedness and simplicity of use (no need for phantom power or batteries). This forms the diaphragm cartridge. each with a particular shape depending on the microphone's design. It won't work very well and you may promptly fry the mic. It is worth noting that the phantom voltage will not harm most dynamic microphones if they are connected to a mixer which has this feature built in . especially guitar. you can connect a raw speaker. violin. a speaker works exactly the same way only in reverse . Certain models are designed to pick up sounds at a distance or groups of people. upright bass. Sound waves cause the top plate to vibrate which alternately compresses and de-compresses the resistance. however they have greatly improved over the years in that regard with many models now designed for this kind of work which virtually equal dynamic mics for road-worthiness. but this backwards PA will actually function (briefly). At one time it was thought that condenser mics were too fragile for PA applications. Other condenser mics are first choice for acoustic instruments.a.nor will the sound be affected. Hence polar patterns are 3-dimensional and really do resemble variously contorted balloons. Sound waves set the diaphragm and coil in motion vibrating back and forth which causes the coil to cut lines of magnetic force. dynamic microphones and speakers are almost interchangeable.k. For what it's worth. Believe it or not. These shapes are what you see listed as "polar patterns" in mic literature. Professional Audio Basics and User Tips Guide 9 .it reacts to the amplified signal by vibrating back and forth to create sound. instruments or instrument amplifiers. "dynamic" and "condenser". The voltage varies in polarity with the frequency of the sound waves and in strength with the amplitude or size of the waves (the louder the sound. however they don't pick all of them up with equal sensitivity. This travels down the cable to the mixer and is amplified.

As a result. Their polar pattern resembles a balloon with the mic right in the center. there are PA applications for them where stage monitors do not get used as a rule including stage plays. It is designed to be attached to a flat surface which effectively bounces sounds from all over the field into the tiny mic's diaphragm. opera.or. The "parabolic" mic. These microphones have been replaced to a degree in live theatre by miniature wireless mics which can be secreted in the costumes. short-wave and CB radios and commercial PA amplifiers as used in factories. hospitals. which is similar in terms of its applications. features a cardioid mic suspended in front of a plastic dish or parabola. This was because PA amplifiers of the day only had high Z inputs and were always situated close to the stage so that mic cable lengths seldom exceeded 20odd feet. to put it in more common terms. all PA mics were high impedance. Shotguns are used to pinpoint distant sound sources and are principally found in the film and television industries. This flat surface can be anything from a sheet of Plexiglas to a wall to the underside of a grand piano lid or the top of a desk. above and below and the sides. in other words they pick up sounds from behind almost as loudly as from in front.• • • • • • Uni-directional or "cardioid" mics can pick up a wide spectrum of frequencies over roughly a 120 to 150-degree sound field.e. However. hotels. stage plays and meetings. Their polar pattern resembles the aforesaid balloon with the head of the mic pressed against one end. Both mics usually have variable directivity . The boundary mic's best applications include grand piano. High Z mics are now principally used with home entertainment equipment. "Omni-directional" mics accept a wide range of frequencies from virtually a 360-degree. but they are still required for film and TV applications and those stage shows where costumes are sparse. etc. aiming into the centre of it. choirs. they exhibit more "directionality".k. these are always condenser mics using a long interference tube which cancels sounds from the sides. In practice this means that they tend to largely ignore sounds approaching from behind making them preferable for vocals and general PA use because they do not readily feed back into stage monitors. "Shotgun" microphones have a very narrow sound field reflecting a "hyper-cardioid/lobar" polar pattern which usually resembles a small. Parabolic mics also must be aimed at the sound source. The "boundary" or "plate" mic offers a 180-degree. Unlike the others. "Z")? Once upon a time.their "pinpoint" factors. They are usually condenser-type. It wasn't until after "Woodstock" and the birth of concert PA technology that there was an increasing demand for mic cables long enough to warrant the need for low Z technology. horribly contorted balloon with the mic partly pushed into it. acoustic instrument groups. In shotgun mics with this feature it takes the form of a small control or switch. In parabolic mics you adjust the distance between the tip of the mic and the centre of the parabola. As a result they may pick up monitors and thus feed back. resembling a balloon which is flat on one side. i. HIGH OR LOW IMPEDANCE ( a. Possible applications include recording round-table meetings or audience ambience. omni-directional mics are less suitable for high-volume music applications and probably should not be used with monitor speakers at all. "hemispherical" polar pattern. They seldom get used for music PA as they have to be aimed at the source which means someone has to hold and aim them.a. provided the monitors are properly positioned. Omni-directional mics may be dynamic or condenser. Since then low Z has become the PA industry standard. Uni-directional mics may be dynamic or condenser-type. however they can be either dynamic or condenser-type and tend to be less expensive than shotguns. "Hyper-cardioid" and "super-cardioid" mics are included in the uni-directional category and generally offer a somewhat narrower sound field . also choir or chorale performances where soloists need amplification. spherical sound field. Professional Audio Basics and User Tips Guide 10 .

aim directly at a point 1/2-way between the bridge and one of the F-holes [condenser.try reducing the input Gain. ride cymbals. cardioid] • Saxophone . • { TIP. Trim. exclusivity (how many sound sources get picked up) and even bass response in some mics.same as above for individual mics. . but audiences can find them irritating. slightly farther away and at roughly a 45degree angle to their mouths if two singers are on the same mic [dynamic or condenser.}. Distortion is more difficult to cure with the mixer . turn down the low-frequency or bass control on that channel to reduce the problem. [condenser. Some vocalists may like these noises. cardioid] Snare drum. If that doesn't work.see above paragraphs [dynamic or condenser. cardioid]. Wherever possible try to minimize the number of mics in use as each open mic reduces the system gain before feedback by 3dB. cardioid] • Keyboard amp . For example. not the sound hole (too "boomy") and get as close as possible [condenser. cardioid or hyper-cardioid] • Professional Audio Basics and User Tips Guide 11 . Handheld vocal mics.usually this is a job for two condenser cardioid mics placed inside. Here are a few standard microphone placement suggestions: • Lead vocals . On a crowded stage with the band playing very loudly. cardioid] • Acoustic guitar . for example.the old one may be fatigued.as close as possible over the sources. When you have a chance.distance the mic a foot or so away to avoid overloading and aim directly at the bell [dynamic.distance and angle. one pointing down at the middle of the bass string section above the hammers and the other over the high strings. Some vocalists prefer that sound even though it may be accompanied by "pops" and other sound effects including distortion.PLACEMENT There are two variables to consider when placing a microphone relative to the sound source . try replacing the distorting mic's diaphragm .If a microphone is creating "thumps" or "pops". certain mics produce a distinct increase in bass response due to proximity effect. • Guitar or bass amp . In a rock band situation. Five or six mics including the one for bass drum should suffice for most standard sets. etc. Distance is the most important factor as it determines a variety of things including signal level. • Grand piano . should generally be kept around 5 inches from the mouth. trombone. A softer. aiming down at the tops. Angle determines tone. this distance may have to shrink to just an inch or two so that the mic will pick up a predominance of that performer's voice or instrument.place the mic roughly 3 inches above the bell and angle it slightly [dynamic. • Bass drum .these usually have a 2-way speaker system which makes miking them complicated.place the mic close to the grill aiming directly at the centre of the speaker [dynamic. clarity. Generally you will obtain the brightest tone with the mic aimed directly at the sound source. mellower tone can be achieved by angling the mic in relation to the source. keep the lid closed as much as possible and experiment with mic positioning (the pianist will have to hear himself through the monitors). but your best bet is to take a "line-level" output from the amp and run it directly into a Line input on the mixer (see >>LINE under THE MIXER. Another approach is to affix a plate or boundary mic to the underside of the lid. • Acoustic upright bass . At this distance however. cardioid]. • Trumpet. cardioid] • Harmony vocals . You could aim two dynamic cardioids directly at the woofer and tweeter then adjust the two mixer channel levels for low and high frequency balance. cardioid]. somewhat closer to the high strings than the bass strings and above the hammers. one for the hihat cymbals and one for the snare.at a distance of roughly 4 to 6 inches. MIXER INPUTS). try using one or two condenser mics over the tenor toms and ride cymbals plus one over the floor toms. Attenuation or Pad setting on that channel.aim directly at the bridge.close to the front drum head or preferably inside the drum [dynamic. tenor and floor toms . try replacing the mic with one which has a little lower sensitivity.

a wise choice). Here is a list of line-level sources and the outputs to use: • Tape deck or CD player . should not be connected to a mixer at all unless first put through a "direct box" with a speaker-level input and a linelevel output. linelevel signals are much less powerful than the output of an amplifier. but the signal is much weaker than the output of a regular amplifier so you may find that a direct box attenuates (reduces) the signal too much. even the tiny variety which powers headphones." output. "Sub"."Line" output. "Mon" or "Send" output. However. • Another mixer .SOURCES LINE-LEVEL SOURCES Anything which provides an audio signal of at least three-quarters of a volt (775 millivolts) can be considered a line-level source."Line" or "Aux. see "LINE" under MIXER INPUTS." output • Keyboard instrument ."Aux. • Instrument amp . It is important to keep these facts in mind because you may be connecting a wide variety of sources to the mixer and all of them will require specific input gain adjustments.". Professional Audio Basics and User Tips Guide 12 . "Headphone" outputs have two problems. "Speaker" outputs however. first they are speaker-level which means that they may distort the mixer input. For information regarding line-level connections. Such a tiny amount of electricity is still several times greater than anything a microphone can generate and greater again than the output of the average guitar pickup."Main". The other problem is that headphone outputs are usually stereo so you will either need a stereo direct box and a mixer with stereo inputs or a stereo-to-dual-mono adapter so that the signal can be split into left and right mono signals and fed to separate mono direct boxes and then to separate mixer channels (or you may decide not to use the headphone jack as a line output . "Aux.

offer greater degrees of protection from hum and noise. LINE" Balanced line inputs. And you may find a phantom switch near the Mic input or somewhere in the master section. The exception would be the output of a "phantom" power supply which is mic-level. in a situation where the output is unbalanced but the input is balanced. but the wire contains two centre leads plus shielding and the 1/4-inch plugs will be stereo. i. The channel input gain control or input attenuator on mixers with such features. Yorkville model LHNT-1). Simply use a balanced patch cable. The standard lead designation is tip=in phase. It's worth noting that.INPUTS "MIC" The microphone input is usually low-impedance. The sections break down as follows: THE MIXER .773 Volts rms. ring=reverse phase. a "low-to-high Z" adapter can be used to overcome the problem (e. 5000 ohms or less in mixers with active input circuitry or approximately 600 ohms in mixers with input transformers (Note: there is no problem with plugging a 600 ohm or lower impedance microphone into a 5.are usually designated pin 1=ground. Mic inputs may also be 1/4-inch "phone" jacks in older or smaller mixers.48 Volts. The term "line-level" is used to cover signals referenced to 0dBm or 0. This travels backwards up the cable to the mic (the only time electrons flow "upstream") for the purpose of powering a condenser mic. You may need to set the Input Gain control higher for high Z mics than line-level sources but some mixers have Mic/Line selector switches so that the Gain settings can be roughly the same for mics and line-level sources.e. Additionally. The channel input gain control or input attenuator on most mixers will regulate the "line" inputs and the input circuits have sufficient gain for high-impedance microphones.usually 24 . Also.000 ohm input). Professional Audio Basics and User Tips Guide 13 . They help minimize hum and noise when connecting line outputs to line inputs. when connected to balanced outputs. pin 2=in phase. If you have a low Z mic and only high Z inputs are available. sheild=ground (although there may be a few exceptions). In this case you use balanced shielded cables which are outwardly similar to regular shielded patch cords.e. This feature applies a small amount of DC voltage to one of the Mic input connector leads . A low Z mic in a high Z input can produce similar results.THE MIXER The various features of an audio mixer can be broken down into several sections in order to simplify explanations. remember to use shielded cables with a centre lead wrapped in the shield wire (as opposed to speaker cables which only have two regular leads). will always regulate the "mic" input. This is where you would connect a high-impedance mic or the line-level output of a signal source. Some of them have three-pin XLR connectors. i. with two insulator bands near the tip rather than one. Almost. anything which runs on AC or DC power produces this level of signal.mic cables can be used for this . It will usually have a 3-pin "XLR" connector and it will always be a female XLR (male XLR's are always outputs). "LINE" Line inputs are always high-impedance (10. but the majority these days tend to have stereo 1/4-inch jack sockets. however they may be high-impedance (over 5000 ohms) so check the owner's manual.1 Vrms or less.000 ohms or more) and employ 1/4" jacks or sometimes RCA (phono) jacks. These are much greater than those produced by mics or instrument pickups which produce signals down around 0. "BAL. The range of things which produce line-level output signals is very wide. Audiopro AP-series mixers feature special balancing circuitry which provides the noise rejection of a totally balanced circuit even when the signal source's line output is unbalanced. a regular patch cord will usually work (ditto the reverse situation).g. A high impedance ("HI Z") mic may produce a low level or distorted sound when connected to a low impedance ("LOW Z") input. pin 3=reverse phase. Beware of "speaker" outputs or "extension" speaker outputs as even the smallest amplifier produces too much power for a "line" input and will overload the circuitry. XLR-type balanced patch cords .

can be added on. these can be used as stereo auxiliary inputs.in other words. unbalanced patch cords from the Insert jacks to the Line inputs on the other mixer. etc. "EFX RETURN" or "STEREO RETURN" The effects or stereo return jacks are usually employed as part of the effects "loops".If your wiring is like the first example with the tip being "send". Using a special "Y" cable with a stereo 1/4-inch plug branching out to two mono 1/4-inch plugs (e. recording or broadcast."INSERT" The insert jack is usually found amoung the channel input connectors and sometimes among the main and monitor outputs. one for each group of input channels. etc. Effects return jacks are usually unbalanced 1/4-inch jack sockets. directly into a channel or main/monitor buss. aural exciter. the patch cable wiring code is tip=return. the "send" part of the Insert jack is buffered . • { TIP . Professional Audio Basics and User Tips Guide 14 .a send and a return. It is a 1/4-inch stereo jack socket which is actually two jacks in one . The mixer will usually have an "aux. Now only the signals which are coming in through the Amp or PA In jack can be amplified.and feed it to the mixer's master RETURN controls which regulate its passage to the next master mixer stage.} "TAPE IN" Tape inputs are most often RCA ("phono") connectors and basically represent the same thing as "Aux.reverb. regulated by the channel Gain control. They are usually found amoung the Main. The purpose of an "aux. perhaps for monitors.The main reason for having multiple EFX or Stereo Return input facilities is to permit you to employ multiple effects "systems". ring=send. Monitor. while in others. if you are using an external power amp to drive the Front-Of-House (i. They accept the output of external units ." input is to permit connecting an additional signal source . the wiring goes tip=send. ring=return. assuming the Insert jacks are not being used for patching purposes. On stereo mixers. Sub. compressor/limiter. Alternatively. channel-for-channel. Yorkville model PC-6ISPH). inputs are stereo. perhaps for connecting a keyboard mixer. sleeve=ground. but may or may not be balanced depending on the mixer (AP mixer aux.perhaps a small outboard mixer for keyboards or drums . In most mixers. • { TIP . main) speaker system. unbalanced). Check your owner's manual to find out which one your mixer uses. • { TIP . You could. delay.g. have echo on the vocals but only reverb on the rest of the band (echo on everything can get too 'muddy' sounding at high volumes)} "AMP IN" or "PA IN" Powered mixers often have a direct input to the built-in power amp. you can patch the mixer's Monitor output into the Amp/PA In jack and simply connect your monitors to the powered mixer's speaker outputs.} "AUX" Located in the master section. In some cases.This means that. sleeve=ground. connectors and will have a Tape In level control among the masters. for instance. they would be in left/right pairs in order to accept the output of stereo effects units. This unusual setup is used in place of two separate jacks to save space (some mixers have separate send & return Insert jacks). you can patch a graphic EQ. digital delay.e. etc. You would simply run shielded. the Auxiliary inputs are always line level and often in stereo pairs. this means that a second mixer." inputs. ." control in the master section to regulate this input and the signal's final destination will be the "main' mixer channel or one of the "sub" master channels. This is a special 1/4-inch jack socket with a built-in switch that interrupts the flow of all the internal mixer signals to the power amp as soon as you plug into it.without using up one or two input channels.

Stereo mixers have two which would be used to feed a stereo F. the "Sub" or "Group" channels can be used as submixes . these outputs are often used in conjunction with Sub or Group In jacks to create loops with EQ's. THE MIXER . usually with its own master and is normally a balanced output (see INPUTS "BAL LINE" for cable wiring). Now the channel Mon.H.house) amp/speaker system. etc. This is included to enable the mixing technician to address the lighting technician. it can be used for a variety of purposes including main PA. sum or mono output represents a mono signal. etc. etc.} Professional Audio Basics and User Tips Guide 15 . In a multi-buss mixer there would. a stage hand or road manager through a small amp/speaker system. "monitor buss".).H.O. Sub 1 for drums and Sub 2 for the rest of the band in a stereo mixer.). instruments or program sources being sent to each buss. or it can be connected to a remote amp and speaker system. "MON" The monitor outputs. If you do decide to go with a mono F.Alternatively. Some mixers may have the ability to also send some of this signal to one or more of the Aux. Expect to see more terms such as "effects buss". of course. similarly numbered masters. which would be dedicated to the needs of the vocals. "SUM" or "MONO" The main. for example. Sum or Mono buss and gets mixed (summed) with the others. LINE for cable wiring). effects units.If you have a stereo or multi-buss mixer. or Line inputs of a cassette deck with a "Y" adapter so that the signal gets onto both tape tracks. The Main. "Buss" is being used here to represent any mixer circuit which receives signals from the input channels. "main buss". be more submix possibilities. drum and keyboard submixes plus one for the rest of the band. like the Main and Sub/Group outs. The various drum channels' Pan controls would simply be turned all the way Left. a mono main mix can work as well or better than stereo in some places. female XLR mic connector. As an example.O. In a multi-buss mixer where there are more than two sub/group channels. • { TIP .H./ Monitor busses so that the performers may be addressed through the stage monitors. • { TIP . Since stereo separation does not work as well for PA as it does at home (people at the sides of the stage may end up hearing only guitar and no vocals. a four-buss board could accommodate separate vocal. and a line-level output. part of the output signal from each Sub or Group buss (channel) goes to the Main.OUTPUTS "MAIN". (front-of. and the rest turned Right. PA."TALKBACK" Some mixers feature a very simple channel consisting of a mic input. Tape or Aux. amp/speaker system and they are balanced as a rule (see INPUTS "BAL. These would be connected to the monitor amp/speaker system. the Main output is exactly what the name implies and would be connected to the main PA amp/speaker system. are usually balanced (see INPUTS "BAL LINE" for cable wiring) and have their own masters. Sum or Mono output is to feed the main F. compressor/limiters. The Talkback input is normally a standard.in other words. perhaps for the control booth or for recording purposes (remember to split the signal with a "Y" adapter to get it on both tape tracks). therefore in this writing it will be reserved for the input channels only. On stereo or multi-buss mixers. it represents the sum of the "Sub" or "Group" buss outputs . one of the possible uses for a Main. a level control.send controls and Monitor master can be used to provide a recording mix while the rest of the board is being used for PA. Now the Sub or Group One master would regulate all the drum mics at once and the Sub or Group Two master would regulate the rest of the band. possibly an on/off button. one of the monitor outputs could be connected to the Aux.O.(Use of the word 'channel' can get confusing when discussing mixers. Mono or Sum Master would now regulate the overall level through the PA. inputs can be used to connect the outputs of the deck to the mixer for playback listening.say. In stereo mixers. In older mono mixers.} "SUB or GROUP" The submix or group outputs are regulated by their own.

Now it will act rather like an automatic volume controller. And. Professional Audio Basics and User Tips Guide 16 . compressor/limiters or other signal processing units which will now be dedicated to those channels being sent to each Aux.If the Tape outputs on your mixer don't have their own master. the channel Level may have to be set much higher than the rest of the channels and the signal-to-noise ratio on that channel will be less than ideal.CONTROLS For the sake of clarity. (On "box". "TRIM". channel controls will be covered from the top to the bottom of the input strip. compressors. chances are the Main or Sub stereo masters control them. While this is ideal for reverb or echo. see under "Processors".Do not connect equalizers. THE MIXER . you simply won't get the extra hum & noise cancellation that balancing provides (you may not hear the difference). "ATTEN" or "PAD" All sources put out different amounts of signal. if there is too little. facilities may not have "monitor" facilities at all.} "TALKBACK" This is the output of the Talkback channel. It is thus important in larger mixers to be able to adjust the amount of source signal entering the channel. Some may be weak and others quite strong. In jacks. by combining them with Aux. not in series with them. On the other hand. have their own masters. (INPUT CHANNELS) "GAIN".} "EFX SEND" The effects send jack is almost always 1/4-inch and may be balanced or unbalanced . one of the Aux. If there is too much. however remember that a regular (unbalanced) shielded patch cord will almost always work with a balanced output or input. etc. What finally comes out is a mix of straight and effects-signals. ART's model SC-2) between the Tape outs and the deck's inputs and set it for "soft knee" compression. this is usually reversed because the inputs are at the bottom of the mixer panel). If there is no such control. compressors or crossovers to the EFX SEND jack. • { TIP . Also. in fact mixers with full Aux. check the owner's manual. You might want to have someone watch the tape deck's meters and counter adjust its record level control(s). like the others mentioned above. A Tape Send or Tape Out control may be among the masters to regulate the level. outputs can be used for recording (see above).. buss. See under "Mixer Inputs" for more details. The effects buss in any mixer is always in parallel with the main busses. As a rule."AUX" The auxiliary outputs are also usually balanced (see INPUTS "BAL LINE" for cable wiring) and. you can create separate "loops" with EQ's. It will be Line level and may be balanced. like a Mon output. • { TIP . An alternative might be to connect a compressor/limiter (e. as stated earlier. For more information.style mixer/amplifiers.g. the rest going straight to the main busses. they're most likely just wired in parallel with the Main or stereo Sub outputs and are regulated by those masters which means that any FOH-system level changes will be reflected in the recording levels. "TAPE" The tape outputs are usually RCA (phono) type and are not balanced. distortion will result. This is also appropriate because signal flow through the channel circuitry is from top to bottom.check the manual to be sure. As a result. only half of the signals go out through the loop. you need to put 100% of the signal through EQ's. they are connected to monitor amp/speaker systems.

all audio circuits. then decrease it slightly. for example. In order to do this there needs to be channel controls to regulate the amount of signal going to each buss./Pad. of course. In order to avoid confusion about how "send" controls work. • { TIP . the channel Clip indicator should be your guide to setting the Gain/Trim/Atten. That way.• { TIP . each channel is capable of sending some of its signal via the internal circuitry to various locations (busses) within the master section. buss is so that you can mix for more than one monitor system." or "AUX. otherwise they will be right after the Gain. This noise gets amplified along with everything else in a mixer. the Monitor and/or Auxiliary send controls may come next. Large concert PA's usually have a separate monitor mixer and someone to run it. but still leaving around 3dB of headroom in most cases (check the manual to be sure). they will come after the EQ section. as a rule. Smaller systems still need to treat the monitor mix as separately from the main mix as possible. the signal should always be much greater than the noise. the best way to mix for monitors is to treat them as a totally independent system. This LED is designed to illuminate when the input signal is approaching the upper limit of the input circuit's capacity. If the signal is very weak.The stage monitors operate in a terribly demanding acoustic environment . some mixers will have a switch marked ".speakers are close to mics and everything tends to be very loud. and the vocalists. the channel Level will have to be increased more than the others thus amplifying the noise so that it ends up competing with the signal instead of being drowned out by it. send control on the channel and more than one Mon.About "signal-to-noise". Ideally. the only equalization they get will be specifically for the stage monitor system. Trim. While a signal is applied to the input. as a result. for example. That is why the channel signals would not. to the huge difference in signal output between mics and CD players. or Aux. A few of the more recent mixers only have a Gain or Trim control. no switch. "MON. even in the most expensive mixers. On mixers with "Pre/Post" EQ selector buttons for these controls. here is a brief explanation.} Professional Audio Basics and User Tips Guide 17 . busses. This is possible because active circuitry replaces the input transformers and. or Aux. The drummer.} At the top of the channel strip." Depending on the mixer design. hence the Gain control is equally as valuable for boosting a weak input signal up to the proper level as it is for reducing the input signal when it's too strong. need to hear themselves very loudly while the guitarist might want to hear a predominance of bass and keyboards because his amp is almost all he can hear. This also is included to help you adjust the input sensitivity for the source signal. they have such large amounts input headroom and Gain control range that a "pad" is not necessary. One way or another. As a result. increase the setting of this feature until the Clip LED flashes.#dB" (minus some number of decibels) or "Pad" . be EQ'd before being sent to the Mon/Aux. The Gain or Trim control provides more of a fine-tune capability in accomplishing this task. The switch will create a very large difference in the signal level corresponding. usually needs to hear himself and the vocals extra loudly. "CLIP" In the absence of a channel VU meter. The reason for there being more than one Mon. the input clipping indicator is your best aid for setting the Gain/Trim/Pad. etc. have a certain level of ambient noise caused by electro-magnetic emanations from all the wiring and equipment nearby. It is thus possible to set the Gain controls simply by watching the channel Clip indicators during a soundcheck and adjusting them for slight amounts of activity. It is not always ideal to have these signals affected by the channel EQ controls since that EQ is there primarily for regulating each channel's sound through the main PA which has different frequency response than the monitor amp/speaker system.

-9dB @ 300Hz (roll off LOW EQ -15dB) Professional Audio Basics and User Tips Guide 18 . are unfamiliar with their function. have to be EQ'd to minimize lowfrequency puffing and thumping sounds as well as feedback. It moves or "sweeps" the MID control's peak or notch in response all the way up to several thousand Hz or down to below one hundred Hz. If needed cut Mid further.e. • { TIP . One application of such a feature is in the fight against feedback. As a result it can have quite a noticeable effect on the sound especially since the MID cut or boost will be interacting with whatever cuts or boosts you may have set with the LOW or HIGH EQ controls.there's a golden rule which says "NEVER OVER-EQUALIZE". The SWEEP control determines what range of frequencies is affected by the MID cut/boost. Harmonica mics. controls may simply be after the EQ (i. As a result.All EQ's function by altering the gain above or below normal over various frequency ranges. first for a boost then for a cut and SWEEP them back and forth. adjust that channel's MID. it could save you headaches later on when the SPL (sound pressure level) eventually goes up and the room acoustics begin changing . for instance. i. that you are using the channel EQ to improve the sound or to get around feedback problems which are exclusive to the FOH PA . For that reason. they are only found on the more upscale PA mixers. Considering that the SWEEP control can alter everything you are accustomed to an EQ doing. Here are some of those tasks & settings: • Killing feedback."EQ" As stated above. As music plays through a channel on the mixer and speakers. a MID boost swept all the way down to the lowest frequency setting will alter the sound of lows AND increase their volume.not always the case.more about that later. resist the temptation to "sweeten" the sound. on tape or for broadcast. the SWEEP will have no effect at all). when it comes to setting the channel EQ . one or more of the Mon/Aux. If the main speaker system has fairly linear frequency response. -6dB @ 200Hz (and roll off LOW EQ -6dB) • "Boomy" bass drum.} EQ "SWEEP" Control Although frequency "sweep" controls have graced the channel EQs of recording mixers for many years. • { TIP . or Mon. Now repeat the process with that channel's LOW and HIGH EQ controls at various settings {but with the volume at a safe level for the speakers}. set MID at -6dB and slowly rotate SWEEP until the feedback stops. This is based on the assumption however. (If there is no MID cut or boost setting. Here you would turn the cut/boost control counter-clockwise to produce a "dip" in the frequency response. the channel equalization is usually desirable only on that portion of the channel signal headed for the FOH system. if it is set at the centre position. Together. Some source signals require basic EQ adjustments to sound "right" whether it's through the mains. As a result many PA users. "post EQ") and are therefore permanently affected by it. it would be worthwhile to spend some time becoming aquatinted with how it works. even veterans. send controls after the EQ with "Pre/Post" selector buttons to put the desired ones through the channel EQ ("Post") or to bypass it ("Pre"). Be careful this doesn't damage your woofers and watch out for your tweeters/horns if you sweep the boost up to the higher settings while the HI EQ is boosted}. This usually comes in the form of one or more cut/boost controls. MID and SWEEP controls can be used to accomplish a variety of tasks from combating feedback to improving the way things sound through the PA or on recording. monitors.If you have set a LOW boost. each with a frequency control to position the cut/boost exactly where you want it along the frequency spectrum.or any EQ . some mixers have Aux. • "Bonky" sounding snare drum.e. then rotate the frequency control until the dip reaches the guilty frequency and the feedback is reduced. This is worth remembering because the way you 'think' things should sound and the way they really should sound to ensure that the system works properly all night are not always the same. -6dB @ 300Hz (with LOW EQ at +6dB & HIGH EQ at +3dB) • "Fwashy" sounding cymbals. Some mixers offer "semi-parametric" EQ. In other mixers.

When mixing effects such as reverb or echo. Most halls. if you have a basic stereo mixer with a "Main" master and corresponding mono output. the Pan control is rotated all the way left..} "PAN" The Pan control. -3dB @400 Hz Floor tom. functions a bit like the "balance" control on a home stereo system. no matter how short the duration. +3dB @ 80Hz (with LOW EQ rolled off -6dB) "Puffing" on harmonica mic. In any case you will learn to use this feature judiciously. Because the PFL/Cue signal is tapped off just before the channel fader (hence "pre-fade") you can shut that channel down through the FOH PA.} "PFL" or "CUE" The Pre-Fade Listen or Cue button sends post-EQ channel signal to the headphone amplifier so that individual channels can be isolated through the phones. In any case.O. perhaps one for drums and the other for the rest of the band.H. With the drum channels panned left (for instance) and all the rest panned right. -9dB @ 80Hz (with LOW EQ rolled off -12dB) Rack Toms. they are affected by both the channel equalization and the channel fader (in PA vernacular they are "postpost").H.a squealing amp. and you are running a mono F. e. There may be more than one EFX send control and they may feed either an internal effects circuit (reverb) or a master Effects Send buss. only the speakers on the left side of the stage will be producing that channel's output . the channel EFX signals are internally routed to their designated master effects summing busses where they are mixed together on their way to the EFX SEND jack or internal effect. system is to get the sonic benefit of a stereo reverb.g. but when you need to solve a problem it's good to know how to use the tools. the effects send controls are always post-EQ and post-fader. etc.} Generally speaking. If. the only real reason for running a stereo F. check the owner's manual if there is more than one EFX control on each channel to see which one is which. (main) PA is stereo.O. the PAN controls can be used to establish two main mixdowns. system. Use them as a starting point and "tune around" them. In fact it regulates how much of the channel's post-EQ signal gets routed to either the Left or Right Main PA busses. -6dB @ 200Hz {Note: These are approximate settings only.. • { TIP . +3dB @ 5kHz (with HI EQ rolled off -9dB) Fading vocal range (notes too low for singer). have at least some natural reverberation and the sound can become ill-defined or "mushy" if just a little too much reverb/echo is applied. However. When you consider that the natural hall reverb is likely to muddy this effect and you aren't likely to be using a lot of reverb anyway. that channel's signal will only go to the left Main buss. you will probably end up with the MID in cut mode for most problem-solving uses of the SWEEP control. "positioning" certain sources in the soundfield .g.• • • • • Excessive hiss from guitar. If the F. Professional Audio Basics and User Tips Guide 19 . clubs.In most PA situations. other than certain recording applications. a distorted mic. you have to wonder what the PAN controls are good for.. This is a convenient feature for previewing channels before bringing them into the mix (e. but still hear it through your headphones. the other becomes the band master and your mono Main fader regulates overall level. etc. don't overdo it. you are probably better off not to use any reverb/echo at all. In places with an audible echo. "EFX" Unlike the MON/AUX send controls. • { TIP . The best PA EQ setting is the one with the LEAST adjustment. found only on mixers with stereo Main outputs.e.H. bass or keyboard amp. for cueing tapes up). the Left submaster fader now becomes the drum submaster. It may also be used for checking out problems . for example. i.O.not an ideal situation.

This feature is most commonly used to combat certain persistent feedback problems where two mics are picking up the same source and feeding it to a nearby speaker. you may employ the stereo submixing section of your choice and Pan between its two masters. "MAIN" This master regulates the output of the MAIN. That way. "SUB" or "GROUP" These masters regulate the SUBmix or GROUP output levels./Aux. pre-recorded music or sound effects.} "PHASE" or "POLARITY" The input phase or polarity reversal button may appear at the bottom or top of the channel. Aside from that. sends then shut the channel off to be added to the music program later on. • { TIP . one exception being the EFX or STEREO RETURN master(s) since those jacks are inputs. • { TIP . Its prime function is to enable the user to pre-set a channel's level. If a mic is feeding back for instance. etc. See under MIXER OUTPUTS for further information on any of these features. EQ.The Mute button is often used as a quick first step to getting rid of a problem. post-pan channel output to selected "pairs" of Main mix busses. if muting the channel does not cure the feedback. Multi-buss mixers often feature pushbuttons on the channels which direct the post-fader. "1-2". all of which should be left off when not in use to reduce unwanted sound pickup and the risk of feedback. usually a monitor. Turn down the Monitor control and if that doesn't work. Muting is a convenient feature for infrequently-used channels such as harmonica mic. We mention it at the end of the channel section simply because that is where the button most often appears . SUM or MONO buss where the outputs of the SUB or GROUP busses or Left & Right stereo master busses get mixed down into a single signal. everything does exactly what its name implies. mandolin.i. certain wind and percussion instruments. return that channel to normal status and try muting the next most suspicious one (nobody said pro sound was going to be easy). As the name implies. banjo.} (MASTER SECTION CONTROLS) The masters generally act as output level controls for their designated output connectors. either the monitor is feeding back or you have the wrong channel. acoustic guitar." AUX" or "EFX" These masters regulate the output level of their designated SEND or OUT jacks. Professional Audio Basics and User Tips Guide 20 . Of course. "L-R". "MON".e. take the channel off Mute and bring the fader level back up. "RTN" These masters regulate the input levels of their designated RETURN jacks. it flips the polarity of the input signal so that it is 180 degrees out of phase with the other channels. then lower the channel fader level. conveniently close to the channel fader and PFL/Cue button. "3-4"."MUTE" The Mute button is usually inserted just after the EQ section. For more details see Running the System item [4]. you can Mute it then EQ the feedback on that channel (see >>EQ above) or have someone move the offending mic. Efx sends and Mon.Remember to de-select one pair of submasters when changing to another to avoid gain buildup and feedback. As the name implies it silences the channel through the FOH system and possibly the monitors (check your manual ).

pull the EQ faders down one at a time. "gain") over various narrow bands of frequencies. but boosts of more than 6dB should be avoided. See under "MIXER INPUTS" for more details. the golden rule is NEVER OVER-EQUALIZE. "TALKBACK" This regulates the level of the Talkback output. but one sends the effects signal to the input of the main busses and the other to the input of the monitor buss. As mentioned earlier. If possible. remembering to push them back up to centre if the feedback doesn't stop. For more information. "PAN" These controls pan the Stereo Return signals between pairs of SUB or GROUP masters. quite often it's the sign of a good system and a wise technician. LEAVE THE FADERS AT CENTRE . If there is a persistent feedback problem requiring large cuts in the EQ settings. As always.} Professional Audio Basics and User Tips Guide 21 . For DJ applications this is less of a stringent limitation. not the whole system. • { TIP .a.To get rid of feedback.there's no shame in a "flat" EQ. "TAPE" or "2-TRACK" This regulates the level of the Tape or 2-Track outputs. move the mic or the speaker to get rid of it. And when in doubt. the next best solution is to ascertain which channel has the problem then insert an external EQ directly into the that channel (see "Insert" under Inputs). Keep them to a maximum of 3dB (preferably less). EQ cuts cost the system valuable decibels of sound pressure. Eventually you should find the one which reduces or stops the feedback. The process of "sweetening" the FOH system’s sound with low and high-frequency EQ boosts should be done with great care in live music applications. come back in through the Efx Return jack and still arrive inside the mixer at the input of the main or monitor buss circuits at the same time as internal signals direct from the channels . electrons actually travel fast enough for some of them to leave the mixer via the Efx Send jack. "EQ" A graphic equalizer is featured on some mixers and a few have more than one. equalizers work by increasing or decreasing the signal strength (a.Believe It Or Not). This way the necessary EQ cuts will only affect whichever channel has the problem. see "EQUALIZATION" under Signal Processors. If the mic or speaker can't be moved.k. In either case. adjust it back up slightly so that the gain isn't overly reduced. in which case one of the graphics will usually be for the monitor buss. the effects signal gets mixed with the straight signals coming directly from the channels (yes."EFX TO MAIN/EFX TO MON" Some mixers with effects busses feature controls which are actually effects return masters. go through perhaps several cables and effects devices. As a result they are equally capable of curing or causing problems and should be treated with care.

hence altering the tone. let's say a trumpet note. But the dust finally began to settle in the mid-1980's when bin/horn stacks started giving way to advanced all-in-one speaker systems with smooth frequency response and superior passive crossovers. In the mid-1970's digital delays made their entrance along with 31-band EQ's. but that's exactly what happens. in the wake of which PA technology took a huge. They all have inputs and outputs so that they can be connected in series with the audio signal. previously astronomical prices on good quality compressor/limiters had come down and continued to do so.SIGNAL PROCESSORS BACKGROUND Once upon a time there were only two signal processors for PA. Aside from that. The wedgeshaped stage monitor was born along with the dreaded term "monitor feedback" (Which is it . Professional Audio Basics and User Tips Guide 22 . GRAPHIC EQ Try to imagine a set of "volume" controls wherein each one only affects a certain segment of the overall 20Hz to 20. Suddenly 4 and 5-channel PA "heads" were being replaced by multi-channel consoles and separate power amplifiers. GENERAL Signal processors all have two things in common: With the exception of passive crossovers. these are the harmonics. active crossovers and bi-amping. This writing will cover the major ones used for PA . One of them sets the air in motion and the vibrating air then sets the other thing(s) in motion.it should be ringing. The same sort of thing happens along a vibrating string or a reed or anything which vibrates to produce sound . the trumpeter's lips can be vibrating at 440Hz to produce middle A. Actually the faders can't turn things up to "10" or down to "0". but ripples in the skin also produce tiny notes at higher frequencies. Speaker "columns" and "cubes" began giving way to jumbo bins with horns stacked on top. depends on the comparative volume levels at which the harmonics of that note occur. Meanwhile. And the list of PA signal processors continues to grow.000Hz range of sound frequencies. This reduced the need for bi-amping and its inherent dangers for the inexperienced. When you adjust an equalizer you increase or decrease the amount of audible emphasis on any harmonics occurring in that range. • To prove this. Now they're available to club-size PA users. place two acoustic guitars close together and tune them identically. two or more things which are physically prone to vibrating very freely at a certain frequency can get each other in motion by "remote control". The smoked drivers.more or less. spring reverb and tape echo. and how far. It's hard to imagine that merely changing the comparative volume levels of various frequency bands could make such huge changes in the way things sound. far enough to have a marked effect on the overall volume level. That' s a graphic equalizer . What's at work here is a phenomenon called sympathetic vibration. Feedback is simply a sound that's gone wild.guitar effects will not be covered. In this case.it was truly something to behold.the main stacks or the monitors?!). Basically. faltering step forward. the blown amps. depending on how many are pushed up or down from centre. Harmonics are incidental vibrations which occur whenever an instrument or voice is producing a note. Both suffered from mechanical frailty and the echoes had a tendency to overload with amazing ease. The actual tone which identifies something. up until that fateful summer of 1969 and the Woodstock festival. Power amps became stereo then they got bigger and then people learned about "bridging" them. This remained the case throughout the nineteen-sixties. neither of which were well understood. See >>"LINE" under MIXER INPUTS for more details regarding line level. they are all line-level (so do not connect them to amplifier outputs). but they can turn them up to about "7" and down to about "4". they're all different and most require a certain amount of experimentation to get them working properly. the cursing . Signal processors now included passive crossovers and crude graphic equalizers. Strum one of them then mute the strings and listen to the other one .

{ TIP . BUT now the natural tone has been altered (good news and bad news). but on the job it robs the system of power headroom and muddies the sound. as if it is somehow losing power. deep bass belongs more in the home than at a live sound gig. remember that on some if not most mixers. Professional Audio Basics and User Tips Guide 23 . woofers reproducing these very low frequencies at high sound-pressure levels consume huge amounts of amplifier power which leaves less for the important 60-100Hz range where audiences get treated to the "thumps" they like so much. or both. the parametric most often has rotary controls rather than sliders. As a result the "return" part of it is not regulated by the channel's input Gain control. not the channel Gain. if you increase the EQ's Gain by too much and distort the channel. Below 50Hz you enter the realm of very long wave forms and their ability to set all kinds of things in motion . • { TIP . these problems will make the PA sound weak.} • "HP Filter" The high-pass filter.In a live sound situation. Why have a feature which chops off all that lovely deep bass? Because as a rule. nobody would be able to hear it. In other words. As the name implies. sometimes referred to as a "Subsonic" filter. This may cause feedback problems and sound colouration. generally above 20kHz (20. never over-equalize.g. It may be OK at home. bass strings. They cost your audience no real sonic performance and they offer a valuable insurance policy. it's generally a good idea to activate both filters. Boost the Gain too high and you might overdrive the mixer input. is usually activated by a pushbutton. speakers. What happens is. Additionally. the vibrating sources (e. "LP Filter" Opposite to the high-pass filter. a mic diaphragm and a speaker) reinforce each other in an escalating manner at that frequency and you have feedback.One thing you should always resist is the temptation to create a "smiling" EQ curve. ceilings. Be a little careful with this feature. This can cause the voicecoils to burn or otherwise suffer a premature demise. That's why we say again. the low-pass or "Ultra-Sonic" filter attenuates the very high frequencies.. the oscillation gets amplified and goes to the tweeters or horns where it sits. furniture. heating up the voicecoils which are probably unable to reproduce that frequency and even if they were. As a rule. you should reduce the EQ's Gain. You will find controls for Frequency and Q accompanying each cut/boost control so that you end up with three knobs for each frequency.If things with similar frequency response peaks are connected to the same amplifier system and turned up loudly enough. microphones.000 cycles-per-second). The EQ lets you turn down the system's volume at that frequency which gets rid of the feedback. As the volume level rises during the night.} • INPUTS AND OUTPUTS Most professional PA power amplifiers these days feature balanced inputs which may take the form of 3pin XLR connectors or TRS (tip-ring-sleeve) 1/4-inch jack sockets.bass drums.When connecting the EQ directly to an input channel via the Insert jack. Thus. PARAMETRIC EQ Unlike the graphic EQ. it doesn't let anything below that frequency pass which means it's a lowfrequency cut filter ("high-pass" being the correct techno-jargon for it). These may be generated by a defective processor or synthesizer or even a radio station. { TIP . each channel will have two inputs wired together in parallel so that mixer signal can be fed to other power amplifier inputs via balanced patch cords. it lets all the frequencies higher than a certain frequency "pass". you name it. the Insert jack is post-Gain. walls. This is to avoid problems sometimes caused by high-frequency oscillations above the human hearing range.} • Today's Graphics EQ's have additional features worth noting: • "Gain" This is a fader to let you compensate for an adjusted EQ's natural tendency to turn the volume (actually gain) up or down (see above). This provides you with the ability to pinpoint a range of frequencies and either boost or cut the gain there.

The Q control is especially valuable in this regard because once you have located a feedback frequency range using a medium Q setting. sonically transparent limiter can be set so that the maximum mixer signal the amplifier receives is only sufficient to drive it to the pre-determined output power level. etc. surges when the lighting system dims right down. see >>EQ under MIXER CONTROLS. it will cost you performance at high volume levels. as with the graphic. ergo a high setting represents a high gain-change-per-octave ratio. the cut/boost will have a less noticeable effect on a high Q setting because Q reflects the ratio of gain change to frequency band.a "robot maximum volume controller" if you prefer. See above for details. You would patch the comp/limiter in-between the bass amp and the mixer. causing the metal to fatigue then break (fuses also add current-variable resistance to the speaker circuit which affects the amp's damping factor). the Attack and Release should also be around that setting and the Slope or Ratio at around 4:1 or 6:1. Professional Audio Basics and User Tips Guide 24 .Although you can create very specific frequency boosts. establish the right Threshold setting so that the desired power level is not exceeded according to the watt meter. for instance. will always fail after a certain amount of usage because amplifier output signals cause the elements to flex rather like an inchworm.} COMPRESSOR/LIMITER This type of device has been around even longer than either the graphic or parametric EQ. It is worth noting that a setup like this to cover monitors as well as main speakers has the added benefit of protecting against accidental transients from mics being dropped. around half-way. A watt meter attached to the output of the amp can be used to measure power while test signals are put through the system. set the Q up about half way. e. you can increase the Q setting thereby narrowing the range of cut frequencies so that fewer "innocent" frequencies are affected and the basic sound quality does not suffer needlessly. comp/limited signal up or down to match the rest of the system. cables being yanked out. Attack and Release at low settings.g. Controls usually include: • Threshold (the signal level at which the comp/limiting is triggered) • Attack time (how fast the gain gets turned down when the signal exceeds the Threshold level) • Release time (how fast the gain is allowed to go back up again after the peak is over) • Ratio or Slope (ratio of input level to output level.e. parametric EQ's may feature high and low-pass filters.) • Output level for adjusting the final. the comp/limiter acts rather like an automatic gain reducer . Fuses. Together the three controls create a potentially ideal cure for problems like feedback. "1:1" means no comp/limiting. The Q control enables you to broaden or narrow the band of frequencies to be cut or boosted. Like the graphics. Now "peg" that Threshold setting with a dot marker.. You set the cut/boost for a -6dB cut. If the bassist uses a lot of slaps or a hard picking technique.C. then rotate the frequency control until the feedback stops.The Frequency control is self-explanatory . low-frequency boost may sound great at low volume but. turn-on "snaps" when the mixer goes off then comes back on (somebody tripped over the power cord) or A. As a rule. you should (again) resist the temptation to do so in an effort to "sweeten" the sound. For information about scanning for feedback. the Ratio on 4:1. Perhaps more common is the need to ensure that speakers do not receive more than their maximum rated power. Fine tuning of the comp/limiter's controls should net plenty of bass through the PA with no distortion or major loss of system headroom. the "Line" output from the amp is patched into the Line input on one of the mixer channels). "4:1" means input peaks are being comp/limited to as little as one quarter their original strength. etc. A low-Q. The Level control would simply be adjusted in conjunction with the mixer channel's input Gain control to achieve a small amount of clip light activity on that channel. Created as a means to reduce audio peaks on recordings or broadcasts. • { TIP . however its use for PA applications is still largely limited to the bigger systems. A limiter is the best means by far of providing this insurance.it moves the cut/boost across the audio spectrum. A good. set the Threshold fairly low. With the comp/limiter's Output Level on full. One example of this device's usefulness is in controlling bass guitar transients when the bass amp is being "lined" into the mixer (i. hence a narrow band of frequencies.

• Input. Basic features include the following.9:00 o'clock • *Adding sustain to an instrument.5kHz setting. once you have picked an active crossover. if your horn or tweeter is rated at 100 watts at 1kHz.10:00 o'clock • *Bringing out an instrument from the mix.10:00 o'clock. you still need to know what is really happening when you set this dial.When shopping for an active crossover. usually balanced with a stereo1/4" jack or XLR • Low.not likely to cause a problem in this case.000 cycles-per-second). the active or electronic crossover is normally connected between the mixer and power amps and thus is covered in this section.2:1 to 4:1.. Hence. or half of it if it's an octave lower).e. Attack .g. Release . Mid and High output level controls. e. Slope ..10:00 o'clock. Keep in mind that low frequencies kill horns and tweeters even at low applied power levels.18dB/octave or even 24dB/octave.e. As a result.A few other applications for comp/limiters and settings for them are listed below (from the ART model CS2 manual).i. a setting of "1kHz" actually represents a 750Hz crossover point. the high-frequency output signal level from the crossover will be attenuated by -12dB one octave lower than that.Another good idea. let's say 1.9:00 o'clock. Low and High frequency outputs only on 2-way units • Input Level control. and a 100-watt power amp is connected to a 12dB/octave crossover set at 1kHz. • • { TIP 1 . tweeters and midrange speakers or horns receive the correct ranges of sound frequencies so that the overall sound is balanced and there is no distortion or damaged horn/tweeters receiving frequencies which are too low. Its purpose is to ensure that the amplifiers directly powering woofers.000Hz.10:00 o'clock ELECTRONIC or "Active" CROSSOVER Although it is more associated with speaker systems than signal processors. the component may be receiving output down at 500Hz at a power level of 25 watts . power signals from the amp to the horn or tweeter occurring down at 500 Hz. Release . Slope . while it doesn't mean there is anything wrong with the crossover. will also be attenuated by -12dB or roughly 75%. it's probably a good idea to look at the "fast" ones . Crossovers do this by putting the audio signal through a series of filter stages which separate the lows from the mids and/or highs. This test can be worthwhile because not all crossover frequency controls are very accurate and. Knowing this however.} { TIP 2 . the HF components could be receiving 50 watts down at 500Hz which is dangerous. Mid and High frequency outputs on 3-way models. then Frequency controls and Output Level controls for each range . Crossovers work by reducing the gain of the audio signal above and below certain frequencies. their frequency variability and their ability to control the volume level of the different frequency ranges. Release . 12dB/octave is alright for small. Note: "# o'clock" simply refers to the general control setting • *Fattening drums & percussion. What makes Active crossovers preferable to the passive variety is their lack of phase alignment problems. low-power systems. But if the amp puts out 200 watts. Thus. If it's high by just half an octave. do not over power them and be sure about the crossover frequency.000Hz (1. but faster rolloffs offer greater precision and a much better safety margin for your midrange and/or high-frequency components. when you are bi-amping high-frequency components in a 2-way system. It is IMPORTANT to note that they do this rather gradually.for example. This can be accomplished using a sweep signal generator and a level meter.} Professional Audio Basics and User Tips Guide 25 . is to check the accuracy of the Frequency control. Slope . one octave below 1. you could compensate by turning the dial up half an octave to a 1. Attack .4:1 to 6:1. 12dB-per-octave crossover at. you could end up with distortion or blown components. i.6:1 to max. on a 3-way crossover there would be a "Low/Mid Frequency" control and a "Mid/High Frequency" control plus Low. For example. if you were to set the high/low frequency of a 2-way. or a pink noise generator and a realtime frequency analyzer. Attack . the rate at which they "roll off" the gain being expressed in decibels (dB) per-octave (an octave is either double the frequency in question if it's an octave higher..

never over-equalize. in so doing.performers need clarity with a capital "C" onstage. just remember the rule.You can adjust the crossover's Level controls for smooth frequency response by "ear". These often have a fixed crossover frequency at around 90Hz. REVERB It follows that. There are also 4-way crossovers which would have either LowMid controls and Output. send equal amounts of signal to all power amps. plug the pink noise into a mixer channel with all EQ set flat including the main graphic . or High-Mid controls and Output. possibly the lead vocalist. Crossover's High frequency Output patches to the Input of the power amp (channel) driving the tweeters and/or horns. Although the RTA provides a handy visual guide. Crossover's Low frequency Output patches to the input of the power amp (channel) driving the woofers. During setup when there are no audience members present. reverb can add a nice "dimension" to your sound and even a bit more if it's a stereo reverb and you're running a stereo main PA. Then you would patch the reverb unit into that channel's Insert connection (See >>"INSERT" under MIXER INPUTS).take it easy with the "boom and sizzle" or the system will start to sound muddy when you increase the level later on. and may or may not have a Level control .if not.{ TIP 3 . a condenser microphone with very flat frequency response (that's important) and a digital frequency response display.the system will be roaring like Niagara Falls.max is always max no matter what the tolerances might be}. Just don't over do it.Send going to the reverb's Input and the reverb's Output going to Return. Now set up the mic in the middle of the audience area and adjust the crossover Level controls until the RTA 's response graph display looks as flat as possible. not your living room . And if you are in doubt. it should not be taken too literally. The maximum settings will compensate for potentiometer tolerances . Standard wiring for an active crossover goes as follows: Mixer's Main or Sub Output is patched to the crossover's Input. Crossover's Mid Output on a 3-way unit patches to the Input of the power amp (channel) driving the midrange speakers or horns. Aside from that. In general terms. they are effects devices not sound controllers and will therefore be dealt with briefly. otherwise your system's punch could turn to MUSH). However you may only want a particular type of reverb or echo on one channel. Note that this channel should probably not be put through any additional reverb as the clarity of that channel could be muddled. Now you can adjust the main EQ slightly to reduce the size of response peaks which look big enough to pose a feedback threat. Professional Audio Basics and User Tips Guide 26 . Realtime frequency analyzers are produced by a few different companies and some sell for only a few hundred dollars. but remember that this is a sound job. Additionally. You still need to use common sense and your ears to a certain degree. Equal restraint should be employed when mixing reverb to the monitors . it's worthwhile mentioning that your reverb unit would normally be connected to the mixer's EFX Send and Return jacks . the level would be preset at maximum. however it's a good idea to employ something else as well for reference. they should be mentioned in that overall context. since reverb units are signal processors. There is a basic rule of thumb which says that a small amount of your favorite reverb sound on the main system should be enough for most places. the safest thing to do is to set all level controls on the crossover at maximum and. especially if the place has a natural reverb or echo of its own too much reverb makes the sound muddy and lacking in punch (speaking of which. On the other hand. you should never add more than a small amount of reverb to bass drum or bass guitar. RTA's usually come with a built-in pink noise generator. not even to make an RTA look flat. some crossovers may have a "Sub" Output for connecting the subwoofer power amp (channel).

Instead. In the old days. reflecting the amplifier's ability to repeatedly produce clean peaks which last for at least one complete wavelength. They have virtually flat frequency response so differences in sound are often more imaginary than real. home stereo power amps are perhaps not ideal candidates for the rigors of life in the pro audio world. They usually have two Speaker Outputs wired in parallel and may have two Inputs also wired in parallel. find a mono output on the mixer and use that. As a result. MONO OR STEREO? Mono power amplifiers are something of a vanishing breed these days.e. The sections break down as follows: GENERAL INFORMATION Over the years manufacturers have honed the design of PA power amplifiers to a fine edge. Two ratings which are worth looking for are Continuous Average Power and Burst Average Power. the overall speaker impedance will be 4 ohms (more about this later).POWER AMPLIFIERS The various features of a power amplifier can be broken down into several sections in order to simplify explanations. But connect another 8-ohm speaker to one of the channels and that channel will encounter 4 ohms. as with one channel of a stereo amp. etc. We specify "PA" power amplifiers for a reason however. although some of the ultra-compact amps with "switching" power supplies may not be quite as good at driving subwoofers as the heavier ones with conventional power supplies. however a mono system works as well in most situations (more about this later too). continuous output capability. i. Stereo power amplifiers can be viewed as two mono amps in the same package. Other. and possibly distortion. The first rating will be similar to what an RMS test would net and the second one will be higher.Which Watts are Which? Amplifier power ratings these days tend to be in watts expressed as "continuous average" or "burst average" or "peak" or "music power" or "continuous music power". the only continuing difference between power amplifiers is reliability and even that tends to be less of a variable as designs improve. however.}.perhaps subwoofers. the extra input can be patched to feed some of the (mono) mixer signal to the input of another power amp . If the mixer does not have a mono output. As a result. you can connect two 8-0hm speakers. one representing the sum of the left and right channels. simply pan all the channels left or right and use the output you've panned to. Aside from that. If you do connect a mixer's stereo outputs to these inputs there can be phase cancellations causing a change in the sound. one to each channel. Professional Audio Basics and User Tips Guide 27 . Dual parallel Inputs are not intended to accept stereo mixer signals. channel one can power a (mono) FOH (Front Of House) speaker system while channel two powers monitors or anything else . it's really hard to find a modern amp which doesn't live up its specifications. and the speaker impedance encountered by the amplifier will still be 8 ohms. And. however they do exactly the same job as one channel of a stereo amp in terms of basic sound quality and are still employed in a variety of applications from live PA to installations. if you connect two 8 ohm speakers. POWER .To connect a stereo mixer to a mono power amplifier or one channel of a stereo amp. because each channel can operate quite independently of the other. Hence. there isn't much wrong for a power amp to do. the nomenclature was "RMS" which stands for "root mean square" and reflects the results of a test for the amp's long-term. connected speakers will be in a parallel circuit even though there are two Output jacks. Stereo FOH PA systems are becoming more common these days.rather like using a "Y" adapter. In reality. more modern tests tend to net fairly similar results but are more complex and require more sophisticated equipment. Also remember that. • { TIP .

reflects an amplifier's potential ability to counter or "damp" the effects of this process thus permitting more power to be delivered to the woofers. tube amps have a "favoured" impedance above or below which their maximum power capability decreases]. but anything down to 20 VUsec is also considered good. This means that the amp can deliver double that output on frequent. especially woofers. blow up because the speaker load was too low. This is supposedly because amplifier outputs encounter a certain amount of "Counter-EMF" (electro-magnetic force) from the speakers. in its popular perception. There is some truth to this. Heaven forbid. check the specs for headroom figures. for example. full-wave peaks.HEADROOM A good amp will have 3dB of headroom at its maximum. Damping factor. Professional Audio Basics and User Tips Guide 28 . Once again.30 Volts per microsecond . as the impedance decreases. a solid-state amp is able to put out more power. However the truth is that slew rates have to be unusually low . but the truth is. although an amplifier may have flat frequency response all the way down to 20Hz. SLEW RATE Slew rate is basically a measure of the amplifier's ability to supply voltage in response to fast. an amplifier's output impedance is 0. DAMPING FACTOR Damping factor is usually. As long as the amplifier's output impedance is lower than the overall speaker impedance you have damping.especially if we've had an amp shut down or. In other words. continuous output.01= 400. especially on low-frequency peaks where the cone is travelling very far in and out causing the voicecoil to cut more lines of magnetic force hence generating more counter-EMF voltage. What happens is. That's what really helps to make those deep "thuds" and "rumbles" shake the floor when you are powering subwoofers. it just takes a holiday. continuous output. Conversely. All well and good. That characteristic can be chalked up to large amounts of power "headroom". it won't put out any power at all. the amp actually tries to put out more power than it was designed to deliver. in other words an infinitely high impedance. its ability to make speakers reproduce low frequencies with maximum sound pressure depends to a certain degree on its damping factor.is considered good. This means that the amp can deliver double that output on frequent. The popular thinking goes as follows. amps are not harmed when connected to high load impedances. If.[Tube amps are very different in this regard. and as it increases the opposite happens.more about that in the Speaker section). But we all know that this is not the case . POWER AT IMPEDANCE An amplifier varies its maximum output capability in accordance with the overall speaker impedance it is driving. but not as much as you might think. Solid state amplifiers do this inversely to changes in the speaker impedance (yes. speaker impedances actually change with the frequencies they are reproducing . High frequency reproduction is thought to be better in amplifiers with a high slew rate hence. In fact the method of calculating an amplifier's damping factor is based on the difference between the two impedances.well below 10VUsec before any highs are likely to be muted. And when a solid-state amplifier is connected to no speakers at all. if you do hear a difference between two amps with radically different slew rates it would be in the highs. linked to low frequency reproduction. full-wave peaks.01 Ohms and it is designed to operate into a 4-Ohm impedance. a high damping factor does not guarantee better bass response from the speakers. the net result being a reduction in deep bass response. but not very accurately. their maximum output is simply reduced. shortduration peaks and is measured in volts per microsecond (one millionth of a second). [For what it's worth. It would seem to follow that solid state amplifiers might be able to deliver more than their rated power if you connected them to very low impedances. In fact a really high damping factor can have an effect on the woofers which makes their lowest frequencies roll off more abruptly than if they were being powered by an amp with low damping. A good amp will have 3dB of headroom at its maximum. Damping factor is a matter of impedances. A slew rate of "30VUsec" . Few amps like that are produced today. the damping factor would be 4 / 0. They will self-destruct if run with no load connected]. It overheats in the process and will self-destruct if it has no thermal protection circuitry to shut it down or otherwise limit its activity. These induced signal voltages tend to be out of phase with the amplifier's output and can cancel some of it.

to patch the two channels together. heat from the output transistors is radiated away. The heatsinks. it is critical to those which are passively cooled. however it could be a wiring nightmare. Dust sticking to internal heatsinks can act as an insulator and reduce their thermal transfer. this is a very rough rule because all amplifiers react differently to speaker impedance changes and almost none of them puts out exactly half or double power into double or half the impedance. the output transistors are tightly fastened to metal fin clusters called heatsinks attached to the outside of the amp.k. The chassis (case) of the amp helps to trap the air inside and further ensure that it travels over the fins. fans tend to be thermally regulated so that they either rotate slowly or nor at all when the amp is running cool. Before we leave this subject it's worth mentioning that a rough rule-of-thumb regarding power at impedance goes as follows: double the speaker impedance and you cut the amp's output in half . for example. will it ruin my amp?" Oddly enough the answer is. • { TIP 2 . not necessarily. choking off cool air. Professional Audio Basics and User Tips Guide 29 . the result can be overheating and possibly damage. As air passes over the fins. active cooling is the norm. Passive cooling is common on units rated at under 800 watts. Again.• { TIP . all power amplifiers were passively cooled. • { TIP 1 .} POWER AMPLIFIERS FEATURES & SPECIFICATIONS INPUTS Most PA power amplifiers today offer a selection of input connectors and facilities. You could.} In an actively cooled amp. whose size largely dictates their thermal transfer ability. however. Although it is theoretically possible to passively cool a one-thousand or two-thousand-watt amplifier. usually the B channel input. "If I connect too many speakers. And by making a special series/parallel wiring rig. would have to be huge and so would the amplifier. in power amplifiers over 800 watts. Racks or cases should always be open-backed and placed well away from walls or other obstructions.see below).Clean or replace the air filter regularly. Balanced inputs may take the form of 3-pin XLR's and/or 1/4 inch TRS (tip-ring-sleeve .. If you are in doubt about what an amplifier delivers into a certain impedance. "stereo") jacks. If there is restricted air flow.Although proper ventilation is important to all amplifiers. It may be worthwhile to have a technician clean the internal heatsinks once every two years or so if your amplifier does not have an air filter.Be sure that the overall speaker impedance being driven by each amplifier channel is not lower than the manufacturer's specification. This way. a built-in fan moves the air across compact. you could theoretically run as many speakers as you like from one amplifier.cut the impedance in half and (assuming it's not too low now) you double the amp's output. assuming it has a minimum load rating of 2 ohms per channel. run as many as four 8-ohm speakers from one channel of an amplifier. if it is only a little lower than the amp's minimum rating. the long-term amount of dust drawn in by the fan is reduced. it is impractical.a. If the rating is much too low for your amp to handle it may cause serious problems for your power amplifiers.a. internal heatsinks. you might be alright. PASSIVE & ACTIVE COOLING In the beginning.} A COMMON QUESTION. This will prevent it from getting clogged with dust.. With passive cooling. The amp operates with both channels reproducing the same (mono) input signal until a jack is inserted in the B input which breaks the internal patch so that the channels will be independent. As a rule there will be both and the 1/4 inch jack will be wired parallel with the XLR so that you can patch to the input of another mixer or to the other channel's input on the same stereo power amp (a Stereo/Mono button can make this unnecessary . In most of the newer amplifiers. check the manufacturer's specifications. On the other hand. Some amplifiers such as the old Beta-800 have an input feature which utilizes internal wiring and a special switching jack.

run the gamut of just about everything from 1/4-inch jacks to post (a.until recently. the only question is how and why they should be used. some may not in which case it is necessary to connect a mono mixer signal to both channels either using a "Y" adapter or by patching to the other channel via the additional input if there is one. a. You would then connect half of the subwoofers to one channel of their power amp. This lack of standardization has been a sore point with pro audio users for years. for example.enough to carry no more than a few hundred watts before heat and then resistance begin to build up at the jack tip thus decreasing power delivered to the speakers. There are exceptions of course. hence legislation is looming somewhere in the future to remove binding posts from power amps.a. XLRs. If you're using balanced patch cables. these are not "balanced speaker outputs"). MONO STEREO SWITCH In the "mono" position. Professional Audio Basics and User Tips Guide 30 . over the years. in this case with the power of both channels combined into one output signal. Mono is a common amplifier operating mode for PA applications involving multiple speakers.Be very careful about bridging . something XLR's do not have. to 3-pin XLR connectors (no. Speakon connectors were specifically engineered for the application which their name implies and may be destined to eventually become the standard connector for high-powered amplifier outputs and speakers alike. you would employ two amplifiers in mono mode. you would probably have the subwoofer channel at maximum and the other at a lower setting. for example if one channel is powering the subwoofers and the other the full-range cabinets in a club system. This is accomplished by taking the cable-end apart and resoldering the leads in reverse order. at least the high-powered ones. Most people don't do things like that. reverse the "phase" and "reverse phase" wires. The second problem has to do with user safety.a. for instance. but consumer safety groups usually have their way. XLRs work fine.k.MUCH less than desirable (mic cables just heat up and waste power. Driving manufacturers to make all these changes is the need to accommodate very high output power levels. For most PA applications you would run them at maximum. and the other half to the other channel. • { TIP . If. In either case the other end of the cable will need to have whatever connector matches the ones on the speakers .it can be difficult even for people who are familiar with it. Read the "Bridging" section below. a practice which nets less than desirable results . In an unbalanced line. reason being that you want the amplifier's full output capabilities available. to the "Speakon" locking connectors developed by Neutrik in Switzerland. mono. They even have accommodations for bi-amping built into the basic design. Also in a club PA. you can receive a very nasty shock by touching post terminals while a kilowatt-plus amp is in use. The 1/4-inch jack. LEVEL CONTROLS The function of power amp Level controls is simple enough. Although modern PA amplifiers tend to have a Bridge switch. "banana") terminals. only has around one square millimeter of contact area at the tip . you needed to power four subwoofers and four full-range enclosures. this feature ties the two channels' inputs together via internal wiring so that they operate in unison. people occasionally use the wrong ones to connect speakers. This leaves XLRs and Speakons.k. to combination XLR/1/4-inch connectors. Post terminals carry plenty of power and have been the standard speaker output connection for many years. reverse the hot and ground wires. First. but due to the cables' resemblance to mic lines. Then you reverse the phase of the signal going into the other channel. both of which appear on a few amps at present. assuming they don't melt). however there are two problems with them. one fed by the low frequency output from the electronic crossover and the other from the crossover's high-frequency output. cables have to be either bare-ended and wrapped around the terminals or equipped with "banana" connectors which go onto them.} OUTPUTS Output connections on PA amplifiers have. you may be using a single monitor mix for everyone and thus the amp channel driving the drummer’s monitors could be at max and the other at a lower setting (drummers aren't deaf.The emphasis on designing features into stereo amplifiers which make them easy to convert to what might be called "dual-channel mono" operation further illustrates that this does tend to be the way they are often employed outside the living room. just noisy). "Bridging" (not to be confused with dual-channel mono operation) is another way of turning a stereo amp into a mono amp.

If your amp has post terminal outputs this is easy. If the amp has XLR outputs. once again. • Check the impedance of the speaker (yes. mono) mixer signal to both inputs. • DO NOT ENGAGE THE BRIDGE SWITCH UNLESS YOU HAVE PERFORMED THE ABOVE FUNCTIONS.e. • If the amp can handle that load. follow the same basic procedure as with XLR connectors (except. for example. the Bridge switch only patches the two channels' Inputs together and puts one of them out of phase. the switch bypasses the internal input patching so that the channels function independently. of course. switch it "off". but reverse the phase of the signal going into one of the channels. so that they will be in phase. its input-to-input internal patch wiring has the positive and negative leads reversed so that one of the channels is 180 degrees out of phase with the other. This is not necessary if you have a Bridge switch. only one speaker. each plug connected only by the tip tab (the shorter of the two tabs inside the plug). you would be WASTING power because the speakers connected to them will ALSO be out of phase and CANCEL EACH OTHER OUT acoustically. unsoldering the wires.e. A 4-ohm load.If your system is sounding mysteriously "weak". will seem like 2 ohms to a bridged amplifier and it will react accordingly.The full range speakers would be similarly connected to their amplifier. be sure to note which channel is driving the "+" speaker terminal if you will be bridging another amp and speaker. BRIDGE SWITCH This feature does almost exactly the same thing as the Mono/Stereo switch but with a major difference (in fact. you will need to rig a split speaker cable with two 1/4inch plugs at one end. attach whichever connector is needed to the speaker end of the split cable. If it has 1/4-inch jack socket outputs. especially in bass response. Once again. check the Bridge switch. resoldering and then plugging it in. Note which channel is driving the "+" speaker terminal so that you will be able to rig another bridged amp and speaker the same way and they will be in phase (VERY important. using Speakons). In the "stereo" position. Each plug would then be inserted in a single output jack socket from each channel. Professional Audio Basics and User Tips Guide 31 . it is often combined into a mono/stereo/bridge switch). • Set both level controls at MAXIMUM. That weird. rig a split cable as above but with two XLR ends. If the amplifier has Speakon outputs and it doesn't have a separate "Bridged" output. It does NOT somehow "double your power". In fact. • Check the resulting impedance against the amplifier's minimum load rating. This can cause up to a 3dB loss in sound pressure which is equivalent to LOSING HALF THE POWER from that amplifier. you'll need to do this by taking the plug apart going into the "other" channel's input (i. simply connect the two RED terminals. not the channel directly receiving the mixer signal). only one connector per cable end. • { TIP . weak sound would be caused by acoustic phase cancellations between your speakers. especially for subwoofers). This is accomplished as follows: • Feed the same (i. Check the manufacturer's output code in the amp manual to find out which XLR pin is "+" and solder the cable ends to those terminals in the connectors. This is very important because a bridged amplifier reacts to the connected load at half its value. Then plug a cable-end XLR into each channel's output XLR and connect the speaker at the other end of the cable. with the two channels out of phase and just running speakers in the normal manner.} Bridging is the process of turning a two-channel amplifier into a one-channel amplifier producing the summed output of both channels. In any case. no black ones. connect the speaker as follows: take the "hot" (+) signal only from each channel and connect your speaker so that the cable end from one channel goes to the "+" speaker terminal and the other cable end goes to the other speaker terminal. it takes care of phase reversal automatically. Again. reversing the plug leads. read on) and divide it by two. But if you don't have a Bridge switch. Never connect a load which is lower than that rating. If it's "on" and you haven't performed all the following procedures.

Compressor/limiters are a Godsend (see Signal Processors). you still think that bridging two 1. "Protect". If it is very low or the numbers on a digital meter seem confused. A distorted amplifier signal causes the voicecoil to move back and forth erratically in the magnet gap.} GROUND STRAP Ground loops and their attendant hum can sometimes be traced to a rack of power amplifiers. re-connect all the Ground Straps and check elsewhere. you may have a serious problem. limiters can actually maximize clean power. If that fails to cure or sufficiently reduce the noise. Professional Audio Basics and User Tips Guide 32 . built-in limiter will save you untold speaker failures and cost you absolutely no performance. horn drivers and tweeters. even at applied power levels well below their power ratings. Now check everything including the cables and connectors as well as the cabinets themselves. But they are often triggered by output transistor heat levels and can therefore indicate other conditions such as inadequate cooling due to poor ventilation. Use your volt-ohm meter or multimeter to check the overall load resistance. If the amp is fan-cooled. burning the voicecoils. however check your manual to be sure.If you have an amplifier or amplifiers with this feature. This is no longer a problem. If the problem persists. should make such amps virtually irresistible. If these lights flash briefly on power up. In fact. In amplifiers with built-in limiters.000-watt amps is preferable to using one 2. make sure the fan is turning and the filter is clean. If it does. the result is the same . you might consider seeking psychiatric help. Try the following. A good. it probably is a short. Additionally. If these lights come on while the amp is running. • If disconnecting the speaker lines does not make the trouble light go out. The ability to get this feature built into an amp. the problem could be in the speaker circuit . Clip LED activity can indicate that the limiter is working. Although some speaker systems have fuses and/or circuit breakers to help the drivers survive distortion.read your manual to be sure. • { TIP . They can be set so that the amplifiers never receive enough mixer signal to drive them into distortion. "Clip" LED Most manufacturers set the threshold of their Clip LED's to fire at 3 decibels (dB) below the point of distortion. a non-functioning fan or a clogged fan filter (clean or change these regularly). squared or clipped wave forms contain higher levels of current in the delivered power which effectively raises the heating effect. after all this. "Temp. but the limiting circuitry built into power amplifiers has been optimized specifically for them and is most often sonically transparent. check the ventilation. small amounts of Clip light activity are likely to be acceptable. Some amplifiers are designed to go through a status check when turned on .000-watt amp unbridged." or "Fault" LEDs In most cases these lights indicate problems at the outputs created by low or shorted speaker loads. Do not leave the ground straps off unless you have to.an interrupted gig. it may not indicate a problem. • Disconnect the speaker cables and see if the light goes out. sonically transparent. try lifting the ground straps on all but one of the amps in the rack. dirty heatsink fins. get the amp to a repair shop. As a result. If the amp is in a rack or a case.• And if. The practice of bridging harks back to a time when the most powerful stereo amps only put out around 700 watts in total and had to be bridged in order to get all of it into a speaker. just switch it on and leave it on. sometimes contacting the wall of the gap and becoming damaged. make sure the back is open and away from the wall or any other obstruction. LIMITER SWITCH Distortion is DEATH on woofers. Some people think that limiting has an audible effect which can be true when an external unit has been adjusted the wrong way. If this is the case. do yourself a favour. especially at little or no extra cost.possibly a short circuit somewhere.

As a result. you may inadvertently defeat it by repeatedly trying to reset the power breaker or. Also find out which. at least not for long. silent mass. so treat popped AC breakers and fuses with RESPECT. amplifiers do not "create" power. whether you employ a variac or not. replacing the fuse or resetting the breaker probably won't help. Even speakers with fuses or breakers built in may succumb to DC from a big amplifier. POWER It's worth noting that. not only amplifier problems. But a high current rating does not guarantee that you are going to get sufficient AC Voltage out of the wall to power your amplifiers up to the manufacturer's specifications. But. Then what do you do about it? Not much except perhaps to employ a "variac". The life you spare may be your own} Professional Audio Basics and User Tips Guide 33 . Too bad because they can be invaluable when you begin encountering. that circuit will need to have a 30 ampere ("amp" for short) current rating. • { TIP . highpowered variacs are quite expensive and the good that they do may not be necessary everywhere. it MUST be on a separate AC circuit. a gadget which allows you to adjust the AC voltage upward when it is sagging below 117 or 120 volts. but also oddball misbehavior from your digital processing equipment or keyboards. This is called "DC offset" and it instantly turns delicate voicecoil wire into a black. if any. The problem is. The power that really drives your speakers is what comes out of the wall. some of which require a strict "diet" of not less than 110 VAC in North America. For that reason they are often overlooked as part of small to medium sized PA rigs. or wrapping the blown fuse in foil (argh!). then you adjust the variac's control until its meter reads the desired voltage. If nothing else. It is most likely time for a visit to the shop. distorted and/or overheated amplifiers can be the result of sagging AC line voltages caused by inadequate house wiring. Quite often low sound-pressure levels. You simply plug the variac into the AC outlet then plug your amplifiers into the variac. If you have a lighting system. unless it's a speaker output fuse (some older amps have them) in which case try turning down the volume. what you get out of an amp depends entirely on what goes into it. high-demand electrical appliances on the same circuit as your amplifiers or just too many amplifiers on the same circuit. read the meter's instructions carefully and follow them. you may save an unnecessary trip to the repair shop. Of course. especially if the problem persists. or 220 to 240 Volts depending on what country you're in. Heaven forbid. of the AC outlets have high current ratings. trying to hold it in with something. remember to measure the AC line voltage next time things go screwy. if you have two power amplifiers drawing 15 amperes each and you want them to be on the same circuit for common grounding. not every power outlet is likely to be on its own circuit so always ask the proprietor or manager of the venue which ones are which. it's likely that something bad has already happened. For example. motionless.When using any kind of a meter to check high voltage levels. A Volt meter can go a long way toward explaining why things may not be sounding as they should.CIRCUIT BREAKERS AND FUSES By the time a breaker or fuse blows. contrary to what you might think. they only modulate it. One thing which is sure to pop the mains fuse or breaker is blown output transistors and when they short out in the process of blowing they can let all the DC voltage which is stored in the amplifier's filter capacitors go straight out to the speakers. But if it's the AC power fuse or breaker. Although most PA amplifiers have some form of speaker protection built in. POWER vs.

then to advanced. burned voicecoils). These back-and-forth movements occur rapidly and are thus referred to as "vibrations". monitor the amplifier outputs and adjust the input signals accordingly. but which today often gets used in place of "woofer". Starting with "columns" in the early 1960's they progressed to stand-mounted "cubes" in the late sixties. The amplifier's outputs produce electrical signals which vary in polarity according to the frequency of the source signals. full-range PA cabinets. When these reach the voicecoil. to avoid confusion we will not use the word "driver" very much. "sub bin". at any point in time..another word that seems to get used for multiple things . full range bin". We will use the word "enclosure" and describe the type. It is suspended in a circular slot in the centre of the magnet and the outer edge of the cone or diaphragm is attached to the frame. "know thy speaker system". or the sole speaker in a subwoofer enclosure.a word originally representing the compression driver on a midrange or high-frequency horn. The sections break down as follows: THE SPEAKER SYSTEM . Now we face the era of "actively controlled" speaker systems wherein multi-functional electronic processors. have gone through various evolutionary stages.woofers. compression drivers and tweeters . "tweeter" and "midrange speaker".g. e. You might. passively crossed over. encounter no difficulty in boosting the bass EQ for a set of modern. So the bottom line seems to be. etc. users are likely to be employing any one or number of these different speaker systems and they all need to be treated a little differently."mid bin".the sound-producing element in a horn/driver unit. This causes the voicecoil to move back and forth which moves the cone or diaphragm back and forth and that causes variations in air pressure which make our eardrums move back and forth creating the sensation of sound. TERMINOLOGY • "Driver" . • "Compression driver" .a veritable field of study in itself. then to bin/horn "stacks" in the seventies. "mid" (midrange) or "HF" (high-frequency) as the case may be.the low-frequency speaker in a full-range (2-way. a cylindrical voicecoil is attached to the center of the cone or the perimeter of the diaphragm.) enclosure. "all-in-one" enclosures in the mid-1980's to which were added actively crossed over "subwoofers" in the latter 1980's. mid or high frequency "driver" they're asking about.all have four things in common. • "Woofer" . It all works quite simply. designed specifically for given speaker systems. • "Horn" . but try that with an old set of columns or cabinets of lesser quality and the result could be wall-to-wall bits of cone paper (more likely. its electromagnetic polarity varies accordingly which causes it to be attracted to and repelled away from the magnet. for example. BASICS Electro-magnetic transducers . • "Bin" . Now you have to ascertain if it's the low. The problem is. Therefore. 3-way. like mixers. Professional Audio Basics and User Tips Guide 34 . "What kind of driver is in that 3-way enclosure?".BASICS & TERMINOLOGY HISTORY PA speakers. Let's just say for our purposes that this means a complete horn/driver unit. • (1) a voicecoil • (2) a magnet • (3) a cone or diaphragm • (4) a frame or some other structure to hold everything together.SPEAKERS The various features of the speaker system can be broken down into several sections in order to simplify explanations. power amps and processors. This always leads to confusion in conversations.

hence the size of the enclosure and its ports affect impedance.this is an electrical measurement of the physical characteristics of the speaker.a network which separates the low-frequency signals from the high-frequency signals in a 2-way enclosure. Now do it faster. "Port" . To verify your findings. In simpler terms. "Impedance" . Most music represents many sound frequencies from the low (e. "Frequency Response" .a small horn and driver or a high-frequency component which is not attached to a horn. mids and highs in a 3-way enclosure. have various physical factors that cause them to favour certain frequencies. Additionally. It is used to show the interaction between the speaker and amplifier. the fine "sizzle" on top of cymbal sounds which is around 10.the panel at the front of the enclosure to which the woofer. What happens is. That is the woofer's impedance climbing as the signal frequency goes higher. Active crossovers are covered under Signal Processors.• • • • • • • • • "Tweeter" .. the higher the impedance goes and. At 1. look at a woofer's Impedance curve (looks a bit like a frequency response graph in reverse). inductance. it wants to move farther in and out than at other frequencies. Frequency response is plotted on an XY graph with frequency along the horizontal plane and loudness up and down the vertical plane. When a woofer receives an amplifier signal at its resonant frequency. its impedance could be as high as 12 ohms with the amplifier delivering 60 to 70 per-cent less power to it at that frequency.g. down in the low frequencies. The frequencies are expressed in Hz and the variations in loudness are expressed in dB. This is also where you will find the tuning port(s) as a rule. TECHNICAL STUFF!!! . That's impedance at work.IMPEDANCE Hold a piece of stiff cardboard out at arm's length and wave it slowly back and forth. It is measured in decibels or "dB". But in the process of doing so. it can be viewed as the degree of "difficulty" that an amplifier is going to encounter when driving a speaker enclosure at various frequencies. Professional Audio Basics and User Tips Guide 35 . speakers. the less power the amplifier can deliver..Hertz or cycles-per-second represents the rate or frequency at which something (a diaphragm or cone in this case) is vibrating to produce sound.g. It gets translated into impedance in speakers by a few other factors including capacitance.Speakers reproduce different sound frequencies with different amounts of loudness. Although factors in addition to electrical resistance are involved. the woofer's voicecoil cuts added lines of force from the magnet and generates extra "counter-EMF". etc. Even applied power affects impedance. "Crossover" . The higher the frequency goes.000 Hz. If you look at the frequency response graph of a woofer you'll notice considerable "high-frequency rolloff" which reflects the way the line slopes downward throughout the higher frequencies.the speaker box . an 18-inch woofer's impedance may start going up at just a few hundred Hertz. Notice how the air mass is impeding its movement? The same thing happens with cones and even diaphragms to a lesser degree. Tweeter and horn diaphragms suffer somewhat less from this process because their surface areas are small and therefore encounter fewer air molecules. Here we are referring to "passive" crossovers.sound pressure level is the industry-standard measure of loudness. "SPL" .2Hz) to the very high (e. "Hz" . "Enclosure" . the air load of the enclosure (sealed or ported) is the main force on the speaker cone. Woofers. or the lows. narrow "spike" over on the left side.a tall. This raises the impedance whenever the speaker tries to reproduce that specific frequency. Conversely. tend to increase their impedance at higher frequencies. impedance is expressed in ohms. low "E" on a 4-string bass guitar which is 41. You will see how the line begins to curve upward as you look from left to right.one or more holes in the speaker baffle which allow air to move in an out of the enclosure and tune it according to the woofer's resonance in order to optimize low-frequency output.000Hz). as a result. like all things which work by vibrating. tweeter. As mentioned earlier (see Slew Rate & Damping Factor under the Power Amp) counter-electro-magnetic force is the voltage induced in a voicecoil because it is moving back and forth in the magnet's field. complete with its components "Speaker baffle" . reluctance and reactance. for example.mechanical resistance in technical terms. This is physical impedance . are attached. You'll also notice something else on the impedance graph .in this case. This reflects the woofer's natural "resonance".

} THEN. and partly a few other things. As mentioned in the Amplifier section. be sure that you find out the manufacturer's recommended crossover frequency. but this is just an example.g. The solution goes as follows: 1/R = 1/4 + 1/4 + 1/8 + 1/16 = 4/16 + 4/16 + 2/16 + 1/16 = 11/16. then it is measured while operating to verify the load.When shopping for subwoofers. but if high power levels are applied for long enough. In other cases.you need a special wiring harness. • The other reason for needing to know how to calculate parallel loads is because amplifiers don't like running into loads which are too low. the impedance may actually go down within a range of frequencies above the recommended range which would reduce the average impedance and possibly endanger the amplifier. the voicecoil gets even hotter. how do manufacturers determine a speaker enclosure's impedance? First it is designed to operate at that impedance ON AVERAGE. • { TIP . In fact it's quite difficult to put speaker enclosures in series . right?). two 4-ohm speakers in series = 8 ohms).} PARALLEL LOADS (MORE THAN ONE SPEAKER PER AMPLIFIER CHANNEL) Calculating parallel loads is an important capability for two main reasons. Your first reaction might be to boost the low EQ frequencies slightly which is fine. Needless to say. inductance.. Of course you would never do such a thing because the lower-impedance speakers will get more power and be louder than the others (you figured that one out yourself.. partly the crossover's resistance. • first. it could be due to this heating effect.Normally the woofer's magnet acts as a heatsink for the voicecoil. but watch out for signs of amplifier clipping. etc. THE FORMULA • 1/R = 1/R1 + 1/R2 + 1/R3 + etc. But the truth is that everything gets put in parallel. This is important because the impedance of a subwoofer tends to go up quickly when it receives signals higher than those in its recommended range. One thing impedance is NOT is "fixed". This increases its resistance then up goes the impedance and down goes the applied power. Professional Audio Basics and User Tips Guide 36 . ("R" = ohms) EXAMPLES Say you have two 4-ohm enclosures. because dual speaker connections whether on an amplifier. it's partly the electrical resistance of the woofer's voicecoil plus the speaker cable. Others think that if you run a speaker cable from one cabinet to another you put the cabinets in "series" and that just adds the two loads together (e. • { TIP . they will usually shut down if the load is too low and some of them may actually sustain damage. The only time an "8ohm" enclosure is likely to register exactly 8 ohms on a meter is either going to be when the meter puts a little DC (battery) current through it and the enclosure turns out to have that exact DC resistance (very uncommon) or in those instants when the audio program (music?) produces the frequencies which cause the speaker to operate at exactly 8 ohms. a mixer/amplifier or a speaker enclosure are all wired in parallel. an 8-ohm enclosure and a 16-ohm enclosure all in parallel.4545 ohms. Have a technician check it out or you can do it yourself if you have a pink noise generator and a good quality realtime frequency analyzer. Therefore R/(1) = 16/11 = 1. partly the mechanical resistance of the woofer's cone operating within the enclosure and/or horn. as a result. As a result the average impedance will be higher than you would expect and the amplifier will put out less power. it's wise to make sure that your electronic crossover is working at the right frequency (most of them have Frequency controls which are not very accurate). Some people think that if you run separate speaker cables from each speaker output on the amp or mixer/amp to the enclosures you somehow "avoid" putting the speakers in a parallel circuit. the voicecoil warms up the magnet and.If you notice the bass response sounding a little weaker as time passes during a performance. WHAT IS IMPEDANCE? Well. Then. partly counterEMF generated by the woofer's voicecoil moving in the magnet's field.

the better. is a real factor in determining a system's overall impedance.6875 Therefore R/(1) = 1/. We suggest that you print out the "CABLE LOSS CHART" and spend some time studying it. Cable presents the overall load with additional resistance. This interference then goes back into the amplifier's grounding. two 8-ohm loads in parallel = 8/2 = 4 ohms. just get out a calculator and turn everything into decimal equivalents as follows: 1/R = . The effect that it has on the system's impedance is fairly easy to chart out. the better. • { TIP . In general. If you do this it's all just a matter of dividing that impedance by the number of speakers. is the main additive and it varies in direct proportion to the gauge and length of the wire. Similarly.25 + . most people put enclosures of the same impedance in a parallel circuit. the less wire. four 16-ohm loads in parallel = 16/4 = 4 ohms. the higher the loss. It is even possible for speaker cable to add inductance.} Resistance. The following is a quick reference listing of some commonly used parallel loads: ("R" = ohms) • • • • • • • • • 2 x 16R loads = 8R 2 x 8R loads = 4R 2 x 4R loads = 2R 3 x 16R loads = 5.CABLE ISSUES Speaker cable.6875 = 1.67 R 3 x 4R loads = 1.25 + .125 + . Once in a while.If you're not into finding "lowest common denominators". it makes sense to pay attention to what speaker cable you are using.4545.0625 = . is amplified and comes out the speaker system. however. powerful radio signals effectively get picked up by the speaker lines. This could dramatically improve your system performance!! Professional Audio Basics and User Tips Guide 37 . The heavier the gauge.3R 4 x 16R loads = 4R 4 x 8R loads = 2R 4 x 4R loads = 1R THE SPEAKER SYSTEM .33R 3 x 8R loads = 2. For obvious reasons. To keep life as simple as possible. as mentioned earlier. As you will notice in the below "CABLE LOSS CHART" the lower the speaker load. not only according to the length and gauge of wire. In a few instances it has been reported that winding the cables into coils and taping them in place actually helped to reduce the problem. what we offer instead is the effect that cable has on the amplifier's delivered power. but also according to the speaker impedance. Example.In certain rare cases you might find that coiling a speaker cable the right way can get rid of radio frequency interference. The reason we do this is because the percentage of power loss varies.

) 25 25 25 25 50 50 50 50 100 100 100 100 SPEAKER LOAD (OHMS) 8 OHM LOADS 8 8 8 8 8 8 8 8 8 8 8 8 POWER LOSS (% OF WATTS) 6.39 3.98 19.82 4.41 5.35 2.21 Professional Audio Basics and User Tips Guide 38 .94 8.44 15.67 12.41 39.29 23.) 25 25 25 25 50 50 50 50 100 100 100 100 SPEAKER LOAD (OHMS) 4 OHM LOADS 4 4 4 4 4 4 4 4 4 4 4 4 POWER LOSS (% OF WATTS) 12.34 6.10 12.41 5.76 10.34 6.44 15.75 1.WIRE GAUGE (AWG) 18 16 14 12 18 16 14 12 18 16 14 12 LENGTH (FT.33 27.41 WIRE GAUGE (AWG) 18 16 14 12 18 16 14 12 18 16 14 12 LENGTH (FT.39 3.9 23.94 8.76 10.

the SPL would still be around 107dB with 50% of the applied power gone (believe it or not!).21 58. believe it or not!). • { TIP . If you apply the maximum power to a speaker it will (should) provide the maximum sound-pressure level specified by the manufacturer. is because of distortion.20kHz".43 33. The problem with this reasoning is that "too loud" is just too vague and most people like things extra loud anyway. even if the power is not distorted and the amp is putting out substantially less than maximum power.34 6.the SPL will be reduced by -3dB reflecting a decrease in perceived loudness of roughly 30%. Now. If the system was producing 110dB. THE DISTORTION ISSUE The main reason for PA users thinking that you "must" apply 100 watts to a 100-watt speaker or 500 watts to a 500-watt speaker. What often happens is somebody increasing the low-frequency EQ below the enclosure's low-frequency limit (if the frequency response specifications say "50Hz .doubling applied power only nets +3dB or roughly a 30 to 40% increase in perceived loudness ..WIRE GAUGE (AWG) 18 16 14 12 18 16 14 12 18 16 14 12 LENGTH (FT. Therefore the reasoning seems to go that if you have the maximum power available to the speaker. And if you think that's unusual. louder and better.33 27. There is some feeling in the consumer market that a power rating means you "must" use that many watts.41 39.10 12. If your amplifiers don't have such a feature but you do have a graphic EQ. This is not true in the case of PA speakers. Additionally it is sometimes assumed that a speaker's power rating directly reflects its sound output. etc.95 45. let's establish that a speaker's power rating represents its maximum power CAPACITY. the woofers are actually trying to do the impossible and reproduce frequencies that are too low for them.24 THE SPEAKER SYSTEM . possibly less depending on who you ask. but how much less? Consider the following. But that's not all. especially the bass. consider the reverse situation .POWER & SPL POWER First. use them. the amplifier's remaining power headroom gets gobbled up in an effort to make the woofers do the impossible and distortion is just around the corner.} Professional Audio Basics and User Tips Guide 39 . Do these things and your system will sound cleaner. This is also not true.more about this later. pull down the EQ faders which are of lower frequencies than the speaker's low-frequency limit.If your power amps have "High-Pass Filters" or "Low Frequency Cut" buttons or "Subsonic Filters". you will have plenty of power "headroom" and you will never turn up the volume far enough to distort the amp and blow the speaker because by then the system will be too loud.) 25 25 25 25 50 50 50 50 100 100 100 100 SPEAKER LOAD (OHMS) 2 OHM LOADS 2 2 2 2 2 2 2 2 2 2 2 2 POWER LOSS (% OF WATTS) 23.76 10. The reason power capacity is specified in speakers is to hopefully prevent them from being blown up by amplifiers which are too powerful for them.44 15. This causes a rapid drop in their power-handling capacity and they blow. If you apply less than that much power you will get less than the maximum SPL. The problem is that a mere 50 watts can blow up a 100-watt speaker if the amplifier is sufficiently distorted (again.08 22. usually 50Hz will be the low-frequency limit). if you reduce applied power by 50%.98 19.

("Close-coupled" means close together. SOUND PRESSURE (spl) Sound pressure is measured in decibels (dB) and represents the real end product of everything covered thus far. They are expressed in watts. either side-by-side or stacked vertically). you might hear someone say. The main reason distortion damages speakers is because somebody wasn't watching the amplifier "clip" indicators. Once upon a time it was "RMS" and now it's "PGM" (program) power. • SPL varies by +3dB if you double the number of similar. when you see "pgm" you know it means "APPLY NO MORE THAN THIS MUCH POWER".in fact any speaker. equally-powered enclosures.Before we leave this topic. keep in mind that a 100-watt distorted amp can blow up a 100-watt speaker even faster than a 50-watt distorted amp. the question is. in-phase enclosures and the total applied power. Two 100-watt speakers powered by the same mono amplifier or one channel of a stereo amp can handle a total of 200 watts (you already knew that. The rating is 100 watts RMS but you can hit it with two times that much power. • SPL varies by +3dB if you double the applied power to a given enclosure. These are said to more "efficient".at least that's how it seemed. Oh yes. close-coupled. turn down the system level. horns and tweeters are capable of producing more sound pressure for the amount of applied power than others. horns and tweeters were blowing up like popcorn and repairmen were the only ones making a profit . In this case." Go around the corner and you might hear someone else say. EFFICIENCY Some woofers. "Oh I know that speaker. "RMS times three. Now." (argh!). what kind of watts. with one watt of power applied. Meanwhile speakers. it's worth noting that applied power is shared by speakers. you can get the same amount of sound pressure from it with only half as much applied power! Differences in efficiency between home stereo and PA enclosures tend to be astronomical. The average level of efficiency in PA enclosures is approximately 100dB at 1 watt at 1 meter with some being as high as 103dB or even higher. close-coupled. Life is simpler and safer. The following is a chart of sound pressure levels as you double the number of similar. that's how much power you sock into that speaker .watch those power amp clip lights! If they look too busy. Hint . however there has yet to be an industry standard specification established which combines the two criteria. Here are some SPL facts: • SPL dissipates -6dB each time you double your distance from the source. • SPL varies by +10dB if you increase applied power by a factor of ten to a given enclosure. sound-pressure level or SPL is loudness. efficiency is assumed to be 100dB at 1 watt at 1 meter. • For example. everyone knows that. The best PA "boxes" combine smooth frequency response with high efficiency. A slight increase in efficiency can mean a great deal to the performance of the system. Professional Audio Basics and User Tips Guide 40 . Even the best home stereo or studio monitor enclosures can be around ten dB less efficient than outwardly similar PA enclosures. right?). the PA enclosure could provide as much sound pressure with one tenth the power applied. In basic terms. in-phase. Back in the 1970's when everybody was learning about sound systems. One of the reasons for the change away from RMS (aside from the fact that it was a technical misnomer) was all the marketplace misunderstandings about translating RMS ratings into applied power.281ft. Efficiency is expressed as the number of decibels measured at a distance of one meter (3. if an enclosure is just 3dB more efficient than another one. POWER RATINGS Finally we have speaker power ratings. Note. "program" power ratings are more reliable when used the right way. Thankfully.) from the enclosure or component.

the Dept. If the front row is 4 metres or 13. Looking at it. • { TIP . get drowned out by the sound of blood rushing in the vessels in our ears.at low sound-pressure levels we have difficulty hearing them compared to all the midrange frequencies. of Labor considers a "safe" SPL to be 90dB for 8 hours a day. away. Under these circumstances it is the mids and highs which need to be optimized most because they contain the most musical information including the vocals. SPL DISSIPATION In the open air with no wind. certain things become apparent to you right away. the need for a "smiling" EQ setting to provide those extra highs and lows is decreased as our hearing response smoothes out. not a frequency response graph (although it's rather like an upside-down frequency response graph of human hearing in point of fact). In the workplace. On the opposite end of the audible sound range. Rock concert PA enclosures can produce better than 130dB SPL's at one metre. If the alignment is bad enough. the waves will be out of phase with each other to some degree and the coupling effect will be lessened. or in a large anechoic chamber. tends to have a 40dB noise "floor" (constant average noise level) caused by people rustling pages. Be careful that they are aligned properly.12 ft. Even the mythical "long throw" horns obey this law. But it is also clear that. That situation may be alright at low levels when there is ample headroom but. the SPL at that distance could be over 118dB. And if you wonder why it is necessary to shout at a rock concert it's because normal human speech averages roughly 70dB while the PA averages 90 to 120dB in various audience areas. Although the -6dB rule varies indoors with the size and physical makeup of the venue. as the SPL goes up. Note that this is a set of "applied power" curves. the EQ should be flattened out to prevent distortion (and remember to cut the EQ below the enclosure's low frequency limit if the system levels are being maxed-out). it's clear why humans enjoy lows and highs on home stereo so much . and is usually more . stack the horns vertically. however this can vary between individuals. The hearing loss resulting from such encounters is usually temporary unless the experience is repeated too frequently (more about hearing loss under RUNNING THE SYSTEM). furnishings.} Professional Audio Basics and User Tips Guide 41 . If you have full-range enclosures with the horns at the top. If their sound travels farther it's because their output or "source" SPL is higher. and outdoors with wind velocity and direction. SPL dissipates -6dB each time you double your distance from the source. the fact remains that sound pressure does diminish over distance at a rate which is seldom less than -6dB as you double your distance from the speaker system. This is good news because "smiling" EQ settings gobble up power headroom in both the amplifiers and the speaker system. that is if only one enclosure is working. but that would have to be in a very silent place. traffic outside. A more normal quiet place.• • • • • • The average ear interprets a 3dB SPL variation as roughly a +30% or . you may need to optimize the SPL of you speaker system so that audience members at the back can hear properly. the public library. Surprisingly.In large venues or outdoors. this rule does not change with horn loading. FLETCHER AND MUNSEN The frequency response of your hearing changes as the SPL goes up and down. The idea with "close coupling" (what we're doing here) is to have the sound waves from the close-coupled sources radiating in unison so that they reinforce each other.20% variation in perceived loudness. The average ear interprets a 10dB SPL variation as roughly a +100% or . Here again. If the sources aren't aligned. etc. First. The quietest sound we can hear is around 12dB. especially around 3kHz where our hearing is most efficient. the maximum SPL we can handle before it becomes painful is around 130dB. etc. air conditioning. some sounds which are quieter than the noise floor are likely to be at least partially drowned out. External sounds occurring at a level below 12dB. To accomplish this.especially in clubs with sound-deadening architecture. stack them "head-to-head" so the horns will be as close together as possible. sound cancellation can take place. moving about or whispering. This was documented by researchers Fletcher and Munsen in the 1940's and is reflected in the famous Fletcher-Munsen Curve shown below. with increases in the SPL.50% variation in perceived loudness.

8 @ 8w = 109dB. However. @ 32m = 95dB. This reinforcement effect helps to keep the PA audible.The following is a chart of sound pressure levels from one enclosure as they dissipate over distance at 6dB per doubled distance: SOURCE SPL (dB at 1 meter) @ DISTANCE(meters) = RESULTING SPL(dB) [Note: 64 meters = apx. 16 @ 16w = 112dB 1 @ 10w = 110dB. equally-powered enclosures. Professional Audio Basics and User Tips Guide 42 . OF ENCLOSURES @ TOTAL APPLIED POWER (WATTS) = RESULTING SPL @ 1METER (theoretically) 1 @ 1w = 100dB. The following is a chart of sound pressure levels as you double the number of similar. 4 @ 4w = 106dB. you were in the audience. @ 4m = 108dB. When there is more than one source of a given set of waves. in-phase enclosures and the total applied power. 4 @ 400w = 126dB. 2 @ 200w = 123dB. Note. i. @64m=94dB 125dB @ 2m = 119dB. Sound travels in waves. 210 feet] 130dB @ 2m = 124dB. 8 @ 800w = 129B.e. 2 @ 20w = 113dB. Consider that if one of the two woofers in an enclosure. @ 4m = 118dB. they can either reinforce each other when their peaks and valleys are synchronized or cancel each other out when they're not. 4 @ 40w = 116dB. efficiency is assumed to be 100dB at 1 watt at 1 meter. But the 120dB source would result in only 84 dB at 64 meters and would be drowned out. for example. that they have the same polarity. @ 8m = 112dB. @ 16m = 106dB. To put this in some perspective.PHASE & FREQUENCY PHASING and POLARITY It is important for all enclosures to be in phase with each other. close-coupled. 16 @ 1600w = 132dB • { TIP . @ 64m = 84dB MULTIPLE ENCLOSURES SPL varies by +3dB if you double the number of similar. NO. @ 32m = 90dB.It is interesting to note that the difference in sound pressure between one enclosure with 100 watts applied and sixteen enclosures with a total of 1600 watts applied is only 12db! In other words. @ 8m = 102dB. @ 16m = 96dB. 8 @ 80w = 119dB. For example audience noise level at a concert can be so high that the PA may be drowned out if its SPL is only a few dB quieter. in-phase. 64 meters (210 feet) from the system and the audience noise level around you was 90dB. If. @ 8m = 107dB. @ 64m = 89dB 120dB @ 2m = 114dB. 16 @ 160w = 122dB 1 @ 100w = 120dB. ("Close-coupled" means close together. the average ear would hear slightly more than a 100% increase in perceived loudness if the two systems were compared in a blindfold listening test at a distance of one meter.} THE SPEAKER SYSTEM . was accidentally wired with reverse polarity to the other one (manufacturers take pains to check for this) the SPL from those woofers could be attenuated by minus 10dB or worse. It is even more important that dual woofers within the same enclosure be electrically in phase. it would be the equivalent of losing 90% of the power applied to them. @ 32m = 100dB. the difference in SPL at a greater distance would be more noticeable. you would probably still be able to hear the PA which puts out 130dB at one meter (see "Source SPL @ Distance = Resulting SPL" above). let's say a box with two fifteen-inch woofers. @ 4m = 113dB. close-coupled enclosures would combine over distance to reinforce the SPL so that the difference between one enclosure and sixteen would be greater than 12dB at a distance from the source. either side-by-side or stacked vertically). Sound wave cancellation can mean no sound or at least a serious decrease in sound-pressure level and usually a change in sound quality. This is because the acoustic output of the sixteen. 2 @ 2w = 103dB. @ 16m = 101dB. close-coupled.

POSITIONING Positioning is important when you need to have similar speakers reinforce each other's outputs. for example. As a result it is possibly a more common cause of phase-related sound problems than reversed polarity. you were to place two different types of subwoofer side by side. the equivalent of losing 50% of your power. • As a matter of good general practice. The net result would be uneven frequency response and probably less overall SPL than if the enclosures were identical. close-couple all the enclosures side-by-side or stacked vertically and face them in the same direction.e.as far out of phase as you can get. Professional Audio Basics and User Tips Guide 43 .. around six feet apart. Face your enclosures towards each other. Take the cable connectors apart. This maximizes SPL by minimizing phase cancellations. Cabinet alignment is critical for proper acoustic phasing. This will cause the woofer cones to move slightly. Still. Most flat speaker wire has one of the leads marked with printing (the industry standard is "printing to positive" . If they move in. If the reading gets higher when the mic approaches the half-way point between cabinets.connector lugs should be soldered to the same cable leads on both ends. The culprits here are matching and positioning. • Another test method requires a microphone and a VU meter (the level meter on your mixer may work if it is fairly precise). there will be much less phase cancellation which leads us to positioning. check the cables first. If. the lows or mids. Well. the problem could be acoustic phase cancellation where only sections of the sound range are out of phase . The clearest indication that a cable is out of phase is when the wire leads are not soldered to exactly the same lugs on both connectors. perhaps on synthesizer. but reinforcement at others due to their differing phase characteristics. Remove the connector panel and reverse the wires on the input connector.centre tab if it's a 1/4" plug). If the cones move out. low note through the PA. PHASE ERRORS Acoustic phase errors can range from a few degrees to over 100. hold the mic between the woofers of the two cabinets and slowly move it back and forth while watching the meter. unsolder.leads of a speaker cable plugged into the enclosure. touch the positive and negative terminals of a 9-volt battery to the + and . that is if they are side-by-side. • If it appears that something is out of phase. Two boxes with one facing off to the side would have a 90 degree phase error. the cable or enclosure wiring is out of phase. The effect is less low-frequency sound pressure than you would expect. If the cable is not wired out of phase. two subwoofers side by side but facing in opposite directions would be 180 degrees out of phase resulting in at least a -3dB effect on SPL. The rule here is simply to make sure that speakers doing similar jobs are similar. the cabinet's input wiring must be reversed.g. the SPL losses are often lessened by sound bouncing off walls. You may want to get some help from your local service technician for this one!! • If you are having a problem with what seems to be low SPL and all the electrical phasing checks out OK. If not. Similar speakers with reverse polarity to each other are said to be "180 degrees" out of phase . While someone plays a single. If it decreases. the + & . In any case. they're out of phase. However. If they are not positioned immediately side-by-side. PHASE TIPS • To test for electrical phasing. they are in phase. For example. the cable and enclosure are in phase. reverse and re-solder the leads on one of the connectors (only). subwoofers positioned in some manner other than close together will suffer losses as a rule. they would tend to encounter cancellations at certain frequencies.Things are not quite so bad when you have two similar enclosures side-by-side and one of them has reverse polarity (speaker cables soldered-up dissimilarly are usually the culprit here) but the SPL can still be attenuated by around minus 6dB which is roughly comparable to a 75% applied power loss. cable manufacturers are somewhat less careful about phasing than speaker manufacturers. floors and ceilings thus softening the effects of cancellations.

Remember. HORN-LOADED vs. Beyond these. Efforts to produce more compact bass horns in the latter 1970's netted low end response which rolled off abruptly below the bass horn's cutoff frequency (all horns have one) giving them a comparatively "hard" sound. If this turns out to be a problem. but horns still must obey nature's laws and roll off quickly below cutoff. then increase the main mixer levels as far as you can without either offending the audience or causing feedback. Ideally the result would be a perfectly straight and horizontal line from left to right meaning that the speaker reproduces all frequencies at the same SPL. configuration and type of an enclosure are amoung its most obvious features. shape. In fact. if there is one. The sound-producing components are at least equally important and the passive crossover. you 'should' be able to hear it). BASS REFLEX ENCLOSURES The era of horn-loaded "bass bins" has all but passed in general application P/A systems. A response graph which is calibrated in 10dB increments may look deceptively smooth compared to some other graph calibrated in 5dB increments. It seems odd that the most glamorous part of a sound system would be so devoid of bells and whistles. input connectors and finish.few though they may be.a HUGE difference when applied to the sound of something (see GRAPHIC EQ under SIGNAL PROCESSORS). Modern uses of low-frequency horn loading in subwoofers have brought some improvements in performance. response graphs indicate all kinds of peaks and dips with the far right and far left ends bending down where the lows and highs roll off. THE SPEAKER SYSTEM – COMPONENTS AND ENCLOSURE FEATURES The size. extended bass response and/or compactness.Some places can be veritable "SPL sponges". In real life. The degree of accuracy of a frequency response graph can be ascertained by noting the "dB" markings down the left side. a difference of 10dB means either double or half the audible loudness . Here then. The reason "gain" is specified here rather than power is because the amount of applied power will vary with changes in the speaker's impedance at various frequencies. (2) Flatten the main EQ (set all faders at centre ) to give the system as much power headroom as possible (3) Make sure that all power amplifier level controls are at maximum (4) Make sure the Input Gain controls on the mixer channels are all set sufficiently high to allow a small amount of input clip indication. Don't be fooled. but there you have it. a phenomenon related to the architecture. some of the very large bass horns developed during the 1930's and '40's remain among the best in terms of lowfrequency performance which possibly indicates that there aren't many secrets left to uncover in this technology. so that there is line-of-sight with the audience (if you can see a speaker clearly. It is important for you to ascertain this dB scale when evaluating a response graph because it indicates the accuracy of the results.} FREQUENCY RESPONSE Frequency response represents the amount of sound pressure that a speaker produces at all frequencies with a fixed amount of signal gain applied. hardware. The added efficiency which bass horns offer comes at the cost of smooth. Response graphs can be calibrated so that every vertical marking is 5 or 10dB louder or quieter than the one above or below it. comes in a close second. This will probably require stacking pairs of full-range enclosures head-to-head so that the horns are close together. building materials and/or furnishings. The results are detected by a microphone with very flat frequency response connected to a computer which plots them on an X-Y graph with frequency along the horizontal axis and SPL up and down the vertical axis. is a list of speaker features . Fixed gain means that the intensity of the test signals is the same from one frequency to the next (at least that much can be regulated). try the following: (1) Position and aim the mid & high frequency horns. Professional Audio Basics and User Tips Guide 44 . other features include cabinet construction materials.

Reflex technology is similarly well known. SPL may be all you need to consider . the next thing to blow is likely to be voicecoils and that gets expensive. The lightbulb may also help to dissipate excess power headed for the mid/highs and should not cause concern if "sighted lighted" . you may find circuit breakers. rent the speakers and try them out on a job. • A circuit breaker or fuse means an added margin of protection for the components. BOX DESIGN SHAPE . it reduces internal standing waves which means smoother frequency response and possibly improved phase coherence. Higher-powered woofers and amplifiers have now made it possible to obtain higher sound pressure levels from reflex-type enclosures and this. • It seems odd to find a lightbulb inside your PA enclosure. There is an optimum size for an enclosure which is dictated by the various design parameters of the woofer. fuses. However there is a potential benefit in this shape . All of these are desirable and for different reasons: • On enclosures rated at 400 watts or more. Professional Audio Basics and User Tips Guide 45 . Aside from the obvious features such as a variety of input connectors. Square The reason for that "wedge-back" trapezoidal enclosure geometry is to facilitate the creation of semicircular arrays where multiple enclosures are arranged closely side-by-side. Check the manufacturer's specifications. the frequency response will be uneven with a "loud spot" over a narrow range of frequencies causing one or two notes to jump out every time they're played. fast power transients (peaks) will not instantly interrupt mid/high output. but resist the temptation to circumvent these safety devices with aluminum foil. PASSIVE CROSSOVERS Modern passive crossover technology is miles ahead of where it was twenty years ago. Its potential advantages are compactness and smooth. but this feature can perform two functions. keep in mind that it is now possible to obtain surprisingly large amounts of full-range SPL from smaller enclosures. nor do all square boxes fare less than favorably in comparison to them. First. This is not automatically true. If it is too big. You may find it inconvenient changing bulbs or fuses or resetting breakers. it could mean more and/or deeper bass response. but that depends entirely on the woofers. resistors and inductor coils. has swept them to prominence in both concert and club applications. On the other hand. input connectors should include Speakons or XLR's (see Outputs under Power Amp Features). so that large. Created initially for "flying" in elevated clusters. square enclosures . Once again. Also. Specifications can be good guidelines. but they only have around one square millimeter of contact area at the tip which means that resistance and heat will build up there when the number of electrons passing through exceeds the number possessed by the atoms in the contact area. they do at least offer some protection from overpowering and possibly even distorted amp signals. a factor which gives the enclosure a "tight". Their shape looks a little horn-like leading sometimes to speculation about them having "longer throw" than similar. it acts as a time-delay fuse. ENCLOSURE SIZE .when properly configured. extended bass response.of course we know better now. tape. but reality is the final word. Look for the "maximum SPL" and frequency response figures.Trapezoidal vs. combined with their superior low-bass response. high-performance compact enclosure can conceivably blow a much larger but inferior box away. Speakon connectors have many times more contact area having been designed especially for speaker usage complete with extra pins to accommodate a bi-amp or tri-amp "snake" (multi-element cable) and a twist-type locking system. Although such measures are not foolproof. trapezoidal enclosures have also found their way onto stages all over and are used in small numbers with great success. etc. the max. If the response numbers are good. As a rule they blow for good reasons and if you defeat them. usually for the mid and high-frequency components. not all trapezoid boxes automatically benefit from this shape. On the other hand. "focussed" sound. 1/4-inch jack sockets are alright for lower-powered applications.it's just doing its job. lightbulbs and even transistors in addition to the usual capacitors. In fact a well-designed.regardless of the enclosure's size. if an enclosure is large in order to properly enclose the woofer or woofers.Is Bigger Better? The size of an enclosure is often assumed to reflect its bass response and acoustic output potential.

and a high-performance PA component should have a big magnet. This is necessary to support a big magnet's weight. ENCLOSURE MATERIALS Chipboard is generally used in the production of home stereo speakers and studio monitors. Check the manufacturer's specs to be sure.*["Fast" in crossover parlance usually means a rate of 18 or even 24db per octave] On the other hand. It is also comparatively heavy and. check with the manufacturer. some are very plain looking with no brand ID. It is durable and internal bracing in key areas can overcome its tendency to vibrate slightly in large enclosures. but inductor coils . Thirteen to eighteen-ply. Some plywood enclosures are also weatherproof . If the magnet isn't big enough.i. resistors and capacitors may not make an impression on you. The presence of inductors in a crossover circuit means that the rolloff slope is at least 12dB per octave. Full passive crossovers on top of this would defeat one of the benefits of bi. non-resonant material works well for midrange and high frequency horns as long as the driver is adequately supported.are more obvious and are worth looking for. Without them the slope will only be 6dB/octave with no rolloff on the woofer which is acceptable for some stereo speakers and studio monitors. Chipboard is cheaper than most grades of plywood (potentially good news price-wise). hence less woofer energy is wasted by being converted to cabinet vibrations. It also does not like getting wet.is the material of choice for PA enclosures. fiberglass or wood horns. there may be brand identification if the components are from a well-known manufacturer. Unfortunately. So.• The presence of transistors.mostly 7-ply . It is comparatively massive and rigid and resonates less as a result. Times are changing. The larger the magnet is. considering that it principally benefits large enclosures (smaller ones have less wood to resonate and work almost as well in 7-ply) multi-ply wood can be something of a back-breaker. there won't be enough flux to overcome the mass of the high-powered voicecoil (you increase voicecoil power capacity by using heavier gauge wire which weighs more) and the woofer's efficiency will be compromised. 2-way and 3-way crossovers afford the components.or tri-amping . Consider as well that not all cast metal components are great performers. This is because they are designed to be bi-amped or triamped using electronic crossovers and separate low (mid) and high-frequency power amps. the only generalization which is likely to be fairly accurate is that woofers rated at 400 watts or more should have cast frames. the more magnetic "flux" (power) it can hold. Professional Audio Basics and User Tips Guide 46 . As well. WOOFERS. such multi-ply wood costs considerably more as do the enclosures in which it is used. eventually developing the "crumblies" in those areas where it has been dropped or hit with something.e. not all good quality components follow these trends. Its prime advantages over wood are lighter weight and the ability to be made into interesting shapes. are seldom made from this material as it needs to be reinforced with wood or thick fiberglass to prevent vibration. Plastic enclosures have the appearance of being weatherproof and some of them are.coils of copper wire . in the final analysis. Foam insulation or some other internal muting and stiffening material is usually found in plastic enclosures. This negates any weight savings and makes the product more expensive. 3/4-inch wood combines chipboard's mass and rigidness with plywood's durability and can make a fairly noticeable improvement in almost any large enclosure's performance compared to one made of 7-ply. high-tech-looking cast cosmetics are surprisingly easy to manufacture. In reality. Massive magnets need the support of cast frames because cast frames are more rigid than stamped frames which may deform in time due to the magnet's weight and the constant vibrations. but lacks the safety factor that *fast. some high-performance "touring" enclosures may only have DC blocking capacitors on the horns and/or tweeters. for example it was once thought that good mid/high horns had to be made of cast metal. but it does not hold up on the road. In fact it turns out that some cast metal horns have nasty resonances and are actually less desirable than plastic. Large enclosures. no passive crossovers (even the very best ones create small amounts of phase distortion). Since then it has been discovered that almost any ridged.again. Plastic has been used for certain small enclosures for many years. partly for acoustic damping and partly because they tend resonate freely and may buzz without it. Three-quarter-inch plywood . especially high-powered subwoofers. HORNS & TWEETERS It is generally thought that good-quality components will have cast metal frames and large magnets.

but the cones will survive dampness.• { TIP . HARDWARE & FLYING HARDWARE Once upon a time. Carson. more as a matter of taste and tradition than practical necessity. You will lose a little efficiency and the sound may be slightly different.both literally and in terms of legal suits . but above all. although not a complete soaking. Further information may be obtained from ATM Fly-Ware 20960 Brant Ave. California. It still does in the case of most PA enclosures. Professional Audio Basics and User Tips Guide 47 . Someone discovered many years ago that you need more than a chain through the handles to safely hang a speaker over people's heads. This proved to be a headache for enough people . Even the beefiestlooking bar handle can work loose with vibrations over time.A. Seek the advice of a rigging professional before flying speaker cabinets. but has been replaced by indoor-outdoor carpeting on most PA cabinets. Tougher paint finishes have only helped the chipping and scratching problems a little so care still needs to be taken. The carpeting wears better and it's fairly easy to clean.that someone began designing metal fasteners and plates.. There are many ways to fly a speaker.You can weatherproof paper speaker cones to a fair degree by spraying them with Scotchguard front and back. the exception being "flown" systems. U. 980101040. hardware meant handles and corner pieces. Painted finishes are still preferred for touring or installation systems.} FINISHES AND COVERINGS Three things have emerged as the favorite finishing materials with most manufacturers. hardened steel eyebolts and other things to aid in the quest for safe "flight".S. you are putting people lives at risk when you fly them. Vinyl or leatherette covering was used by everyone for many years.

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